[Asterisk-Users] One way audio

[administrator tootai]

Rich Adamson a écrit :

>[...]
>
> (Given you mentioned g729, its
>likely you're trying to use sip across dsl broadband.)
>  
>
Yes

>If you have a nat/firewall function between * and the sip phone,
>look there first.
>
No nat/firewall between

> In followup posts, you might mention how your stuff
>is configured and include appropriate portions of your sip.conf.
>  
>
Setup is

<--------- SIP phones -------->-*
<------- SIP softphones ------> | --- asterisk --- internet
<------- IAX softphones ------> |  on nat/firewall
<H323 softphones through GnuGK>-*       box

What I face is that a SIP call to our GW has from time to time the 
behaviour to "loose" audio. Hanging up and retrying can work, but mostly 
we wait or use an IAX GW and try again and then it work. Can also take 
few hours before it work again.

Our provider see the calls like connected with the right codec, and I 
can see them too in * logs.

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

Thanks for your help

-- 
Daniel