[Asterisk-Users] One way audio

[administrator tootai]

Rich Adamson a écrit :

>Inline and somewhat confused....
>  
>
Sorry for that

> [...]
>
>>Setup is
>>
>><--------- SIP phones -------->-*
>><------- SIP softphones ------> | --- asterisk --- internet
>><------- IAX softphones ------> |  on nat/firewall
>><H323 softphones through GnuGK>-*       box
>>    
>>
>
>Is the above "on nat/firewall" supposed to say "no nat/firewall"?
>
>So, what that drawing suggests is that asterisk is routing packets
>between the Internet and the 192.168.10.x network. Is there something
>else involved? If not, then asterisk apparently is your nat box.
>  
>
Yes asterisk _is_ the nat/firewall box. This is the meaning to the above
"on nat/firewall box" I din't use the good words ;-)

>[...]
>Do your log records tend to suggest the IAX link is dropping, a SIP
>link to somewhere, or what?
>  
>
Logs suggest (and show) that call where normally answered. No drop,
nothing. And the status at the end of the call is answered. Remember
that this audio behaviour affect only one party, usually the called party.

> [...]
>
>If I recall your original posting, it was oriented around a sip call
>using g729 dropping connections. How does that relate to the above
>diagram and the sip.conf entry below?
>  
>
asterisk --- internet --- GW SIP --- internet --- landline net phone
(nearest possible from called party)

[sipgateway-gw]
type=peer
secret=MySecret
username=MyUsername
host=sip.provider
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=g729

> 
>  
>
>>This is the sip.conf relevant part from mainphone BT ATA 186 In this 
>>one, prefered codec are G729, Ulaw and Alaw.
>>
>>[106]
>>type=friend                    ; either "friend" (peer+user), "peer" or 
>>"user"
>>context=default
>>qualify=yes
>>username=106                   ; usually matches the [section] title
>>secret=<secret>
>>host=192.168.10.6
>>canreinvite=no                 ; allow RTP voice traffic to bypass Asterisk
>>dtmfmode=rfc2833               ; either RFC2833 or INFO for the BudgeTone
>>disallow=all                   ; need to disallow=all before we can use 
>>allow=
>>allow=ulaw                     ; Note: In user sections the order of codecs
>>allow=alaw                     ; listed with allow= does NOT matter!
>>allow=g729                     ; Pass-thru only unless g729 license obtained
>>allow=gsm                      ; allways allow gsm
>>allow=ilbc
>>    
>>
>
>Why is there a "allow=" with nothing after the "=" in the above.
>  
>
Error from copy/paste. Isn't.

>The "canreinvite=no" comment says "allow RTP" which is backwards. What
>are you actually expecting?
>  
>
That all the RTP traffic as going through asterisk. As I'm behind NAT...

>Since the explanations for the above stuff is a little thin, I'd have
>to guess that you might have one or more internal sip phones that don't
>have "canreinvite=no"
>
Some of EP don't have this value set, which mean the default will be
used. Those EP are defined but not connected  (used only for test).
Today only my ATA 286 is connected

> and at least one Internet sip phone that is
>unknown as to how it is configured. That's reading way between the lines
>and guessing a lot.
>  
>
No Internet sip phone. All connections are coming from my intranet.

Thanks for help me to debug this.
-- 
Daniel