[asterisk-users] Gizmo Dial Out No CALLED PARTY AUDIO??

[Jim Dickenson]

I do not use Gizmo for inbound, only out. I have a register line that  
looks like yours. In addition I have this:

[general]
context=nonesaid
allowguest=no
allowoverlap=yes
allowtransfer=yes
realm=<my system's host name>
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
maxexpiry=3600
minexpiry=60
defaultexpiry=1200
qualifyfreq=60
notifymimetype=text/plain
disallow=all
allow=gsm
allow=ulaw
allow=alaw
mohinterpret=default
mohsuggest=default
language=en
videosupport=yes
callevents=yes
alwaysauthreject=yes
externip=<mypublicIP>
localnet=192.168.0.0/255.255.255.0
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=60
allowsubscribe=yes
callcounter=yes
counteronpeer=yes
registertimeout=20
registerattempts=10
nat=yes
canreinvite=nonat

[gizmo5]
type=peer
host=198.65.166.131
fromdomain=proxy01.sipphone.com
canreinvite=no
dtmfmode=rfc2833
insecure=port,invite
qualify=yes
fromuser=<myusername>
authuser=<myusername>
defaultuser=<myusername>
secret=<mypass>
context=sip-out
disallow=all
allow=ulaw
allow=alaw


Then in extensions.conf I have this:

exten => _9XX.,1,SIPAddHeader(No-Answer: true)
exten => _9XX.,n,Dial(SIP/gizmo5/${EXTEN:1},20)



-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/



On Aug 5, 2009, at 6:02 AM, Rob wrote:

> Yes ... as a matter of fact here is the sip.conf ... obviously  
> private info removed ....
> [general]
> register => 1747xxxxxxx:xxxxxxxxxxxx at proxy01.sipphone.com
> port = 5060
> bindaddr = 192.168.22.5
> context = incoming
> svrlookup=yes
> ;dtmfmode=inband
> allow=all
> externip=76.98.xxx.xxx
> localnet=192.168.22.0/255.255.255.0
>
> [proxy01.sipphone.com]
> nat=yes
> ;type=peer
> type=friend
> context=incoming
> disallow=all
> allow=ulaw
> allow=alaw
> allow=ilbc
> dtmfmode=rfc2833
> host=proxy01.sipphone.com
> fromdomain=proxy01.sipphone.com
> ;insecure=very deprecated; use insecure=port,invite instead
> insecure=port,invite
> qualify=yes
> secret=XXXXXXXXXXXX
> authuser=1747XXXXXXX
> fromuser=1747XXXXXXX
> username=1747XXXXXXX
> canreinvite=no
>
>
> On Wed, Aug 5, 2009 at 6:07 AM, Administrator TOOTAI  
> <admin at tootai.net> wrote:
> Rob a écrit :
> > Hi all,
> >
> Hi
> > I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN  
> for a
> > while and it works fine .... I just added CALL OUT ... I have no  
> problem
> > with call setup ... the called party hears me ... but I can't hear  
> them ....
> > again if the call comes INTO the server both sides work fine.
> >
> >
> Looks like a nat issue: do you have nat=yes and canreinvite=no in your
> sip.conf for Gizmo5?
>
> --
> Daniel
>
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