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:[email protected]
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:[email protected]
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
<[email protected]> 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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_______________________________________________
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Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
