[asterisk-users] Asterisk hangs up call after 20s
Gilles
codecomplete at free.fr
Wed Dec 22 17:29:10 UTC 2010
On Wed, 22 Dec 2010 14:31:32 +0100, Stefan Schmidt <sst at sil.at> wrote:
>you have a typicall nat issue. Asterisk receives messages from the phone
>but cannot send any messages back (thats why it tries to resend the 200
>ok message 6 times).
>
>try setting qualify=yes to your sip peers config to keep the nat port open.
Thanks for the idea, but all users are defined with qualify=yes:
=============
/etc/asterisk> cat sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
srvlookup = yes
;allowexternalinvites=yes
externip=<public IP>
localnet=192.168.0.0/24
;Other IPs can still REGISTER :-/
deny=0.0.0.0/0
permit=<VOSP IP>/255.255.255.255
permit = 192.168.0.0/255.255.255.0
alwaysauthreject=yes
;for safety
context = dummmy
;all RTP packets go through Asterisk
canreinvite=no
;makes no difference: still hangs up
;t1min=500
disallow=all
allow=ulaw
allow=alaw
allow=gsm
register => me:pass at vosp.com
[vosp_outgoing]
type=peer
host=vosp.com
username=me
secret=mysecret
fromuser=me
fromdomain=vosp.com
nat=yes
canreinvite=no
qualify=yes
[vosp_incoming]
type=peer
host=vosp.com
context=from_vosp
nat=yes
canreinvite=no
insecure=port,invite
qualify=yes
;(!) means it's a template
[sets](!)
type=friend
context=my-phones
host=dynamic
qualify=yes
nat=no
[local-xlite](sets)
secret=mysecret
[remote-xlite](sets)
secret=mysecret
;remote extension behind own NAT: nat=yes or nat=no?
;makes no difference : still hangs up
;nat=yes
nat=no
=============
What's weird, is that the remote XLite can successfully call the local
XLite and I get sound both ways, and it's only 20s into the call that
Asterisk decides to give up and hang up (while the remote side still
thinks everything's OK).
I tried SJphone instead of XLite, same result. Could it be some wrong
configuration in Asterisk?
Thank you.
More information about the asterisk-users
mailing list