* You are viewing the archive for July 21st, 2011

Rebooting a Grandstream

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


> —–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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Per-line registration

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?

asterisk’s SDP

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?

Functions not autoloading

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

No, this is not expected behavior.

Asterisk doesn’t like OpenBTS!!!

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@ for seqno 94 (Critical
Response) — See
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@ – no reply to our critical packet (see

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

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

To: ;tag=as7c57c466
Call-ID: 1348333597@
Server: Asterisk PBX
Supported: replaces, timer

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?

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

Prevent Asterisk from setting CALLERID(name) or unsetting CALLERID(name)