Rebooting a Grandstream

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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:

Asterisk Users 4 years ago 1 Answer

Per-line registration

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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?

Asterisk Users 4 years ago 0 Answer

asterisk's SDP

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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?

Asterisk Users 4 years ago 5 Answer

Functions not autoloading

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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.

Asterisk Users 4 years ago 2 Answer

Asterisk doesn't like OpenBTS!!!

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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

Asterisk Users 4 years ago 0 Answer