That works for us with GXP2000′s and GXP2010, but not the later HD series
GXP21XX.

Alec

> —–Original Message—–
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
> Mike Diehl
> Sent: Friday, 22 July 2011 10:50 a.m.
> To: asterisk-users@lists.digium.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:
>
> [grandstream-check-cfg]
> Event=>sys-control
>
> Doesn’t seem to work. Any ideas?
>
> –
>
> Take care and have fun,
> Mike Diehl.
>
> –
> _____________________________________________________________________
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Hi all,

I’m trying to figure out how it is that a couple lines on a given phone, with 3
lines, can qualify as unavailable while the remaining lines can be available.
I’ve got qualify=1000 in my sip.cfg.

Shouldn’t this be an all-or-nothing proposition?

We have a peer (a Sonus Media Gateway), that sends “a=fmtp:101 0-15″
Asterisk sends “0-16″ back, is there anyway to have asterisk send a 0-15?

On 07/21/2011 04:31 AM, –[ UxBoD ]– wrote:
> Since upgrading to 1.8.5.0 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 1.8.3.0 and I do not see any issues in /var/log/asterisk/messages ?

No, this is not expected behavior.

HI list,
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[1221]: chan_sip.c:3622 retrans_pkt: Retransmission
timeout reached on transmission 1348333597@127.0.0.1 for seqno 94 (Critical
Response) — See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Jul 18 11:25:48] WARNING[1221]: chan_sip.c:3651 retrans_pkt: Hanging up
call 1348333597@127.0.0.1 – no reply to our critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

In the SIP Debug, I see always 10 Retransmissions of the same “SIP/2.0 200
Ok” message!!! after that the above “Retransmission timeout” message is
viewed!!!

Retransmitting #10 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK53934;received=127.0.0.
1
From: IMSI208012601160193

>;tag=lkbdg
To: ;tag=as7c57c466
Call-ID: 1348333597@127.0.0.1
CSeq: 94 INVITE
Server: Asterisk PBX 1.8.5.0-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 213

I have established only a one call from my hardphone (connected to OpenBTS)
to my twinkle softphone. but after the call is dropped (T == 32 secondes) by
my softphone and after hanging up my hardphone (T == 60 seconds) I have
received automatically a call from my twinkle softphone!!!
In wireshark trace, I see that OpenBTS is trying to ACK the OK from
Asterisk, but Asterisk doesn’t like it !!!

I have tried to modify the value of the SIP timers, that works only from a
hardphone to a softphone but not from hard to hard. can some one tell us
what’s the definition of t1min and timert1?
t1min=1000
timert1=5000
timerb=32000

Any help will be appreciated.
A.H. Jos,