That works for us with GXP2000's and GXP2010, but not the later HD series
GXP21XX. Alec > -----Original Message-----
> From: email@example.com
> [mailto:firstname.lastname@example.org] On Behalf Of
> Mike Diehl
> Sent: Friday, 22 July 2011 10:50 a.m.
> To: email@example.com
> Subject: [asterisk-users] Rebooting a Grandstream
> Hi all,
> I've got a number of Grandstream phones and I'd like to be
> able to reboot them remotely, as I do my Polycoms...
> I've got this in my sip_notify.cfg:
On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote:
> Since upgrading to 18.104.22.168 I have had to add into modules.conf:
> load => func_callerid.so
> load => func_cdr.so
> otherwise they do not get loaded even though I have set autoload=yes.
> Is this something you would expect as it is different behavior to 22.214.171.124 and I do not see any issues in /var/log/asterisk/messages ? No, this is not expected behavior.
I have succeeded to establish calls using OpenBTS/USRP1/Asterisk. but the
problem is that my cell phone rings, I get 2 way audio but after a few
seconds the call is dropped. In my asterisk log I see this: [Jul 18 11:25:48] WARNING: chan_sip.c:3622 retrans_pkt: Retransmission
timeout reached on transmission firstname.lastname@example.org for seqno 94 (Critical
Response) -- See
Packet timed out after 32000ms with no response
[Jul 18 11:25:48] WARNING: chan_sip.c:3651 retrans_pkt: Hanging up
call email@example.com - no reply to our critical packet (see
First of all, you should disable unused VoIP protocols. Than remove all
guest accounts from used protocols, disable guest unauth access.
Always use strong passwords for accounts, for users on your system.
Passwords shouldn't be eq username. Move port binds on LAN network for
all active services as much as you can (i.e. SHH should be on WAN too I
Use iptables for blocking password bruteforce. Try to install fail2ban
with jails for asterisk, ssh, HTTP and other public services. Then you
can try to install PSAD (port scan autodetect)…