Mike. Yes sip.ld is the firmware.
I wanted to jump in because i saw you had the phantom ringing problem as well.
I am running 3.3.1 and thought upgrading to 3.3.2 would solve that problem did
you still have the problem in 3.3.2? I thought I saw in the release notes for
3.3.2 that was resolved.
Are you by chance using templates (!) In your sip.con? Ive had access denied
errors befor when running as non root.
- Original message -
> I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a flood
> of:
>
> WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL l
What they are talking about is SIP URI dialling. Let say you have
extension 1000 the rings a phone on your system. With allowguest=yes I
would be allowed to dial SIP:/1...@yourdomain.com and assuming the
context defined in your [General] section had access to exten 1000 I
would connect to that
Is there a way to Force the CDR data to be written prior to Hanging up
the channel?
Thanks!!
Jim
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webin
Rate limiting (google) via iptables FTW! Good luck!
- Original message -
>
>
> Alejandro Imass wrote 20.01.2012 18:09:
>
> > I would like to know how
> to block this MF because he makes calls at 1-2 AM
>
> I use this
> construction on my servers
>
> [users]
>
> exten =>
> _XXX,1
I think in your cdr.conf you are looking for the unanswered= directive.
Thanks!!
Jim
- Original message -
> Hi
>
> I'm using 1.8.7.0 with the RealTime architecture.
>
> If a call goes into application Queue and is abandoned by the caller, no
> entry is made in the CDR. Entries are made
I think the wiki may have just missed func_curl. I have a couple 1.8.x machines
with working func_curl. Have you tried to compile it anyway?
Thanks!!
- Original message -
> Hi,
>
> I've seen that function CURL is missing from 1.8 but back in with 10
> (see wiki.asterisk.org).
>
> With
You got me. At first the polycom world was hard to get into. But with a little
effort to understand the configs and the joys of central provisioning the
Polycom are my go to endpoint. Couple the endless configurablity with Polycom
quilaty and I have many happy clients.
As an aside is that what
Does the user Asterisk is running as have access to this device?
Thanks!!
- Original message -
> I have a USB to serial converter attached to my box. pl2303: Prolific
> PL2303 USB to serial adaptor driver
> if I login to the box and send/receive serial commands over this unit it
> w
Agreed. Check the switch for some kind of port security. Most of the time this
would disable the interface if more than one MAC is present but you never know.
Are there blinky lights on the pc?
Also if provisioning via some sort of server check the MAC-boot log that the
pgone uploads.
Good Luc
Hey All,
Odd thing. I am just trying to return the whole date time stamp from a
SMALLDATETIME field in a MS SQL server.
func_odbc.conf = readsql=SELECT DateCreated FROM [REDACTED] WHERE Code
= '${ARG1}'
Problem is I only get the first 15 back from the field. Like so...
Connected to Aster
Perhaps the Monitor CMD is what you are looking for.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor
Good Luck!!
Jim
bilal ghayyad wrote:
> Hi All;
>
> I need to use the recording for the calls, did anyone try this on Asterisk?
> How it works?
>
> By the way: Asterisk support
I work for a large health care company who does all their dealings with
Cisco and the like. Would not even think of an open source solution for
their telephony needs. The whole company had a need for an emergency
notification system that would be triggered by a phone call and start
paging both
What distribution are you using? Below is a tutorial from the ubuntu
site but it should give you the basics of setting up iptables rules. I
have created custom rules for all my servers and the amount of junk
traffic has been dramatically reduced.
Good Luck!!
https://help.ubuntu.com/community/I
14 matches
Mail list logo
