On 10 June 2014 05:27, ortei...@tiscali.it wrote:
Hello there
I'd like to use sip users and peers realtime.
I think I done all I need to get asterisk works fine in realtime:
res_odbc.conf configuration.
extconfig.conf
sippeers = odbc,asterisk,sipclient
sipusers =
Hi
Is there any harm in using res_mysql for some things and res_odbc for
others?
We already use res_mysql for ARA but could do with having CEL logged to
MySQL.
Thanks in Advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660
I had already done this, but nothing is changed
Thanks
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Ishfaq Malik
Inviato: martedì 10 giugno 2014 10:05
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re:
Ok Ish,
I will try with res_mysql.
Still thanks
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Ishfaq Malik
Inviato: martedì 10 giugno 2014 12:05
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto:
Hi Pietro
That wasn't a response to you but a genuine question for myself out to the
users list!
Regards
Ish
On 10 June 2014 13:13, ortei...@tiscali.it wrote:
Ok Ish,
I will try with res_mysql.
Still thanks
*Da:* asterisk-users-boun...@lists.digium.com [mailto:
Hi
We have the following test .call file and test dialplan:
I can't set a custom CDR var to a value on one channel leg, and another
value on the connected channel leg?
Is there a way I can woraround this issue?
## test call file
Channel: Local/queue@TiagoGeada
CallerID:
As far as I know, only way to set variables on another channel would be:
asterisk -rx core show help dialplan set chanvar
Usage: dialplan set chanvar channel varname value
Set channel variable varname to value
On 10 June 2014 16:39, Tiago Geada tiago.ge...@gmail.com wrote:
Hi
We
Using Set(MASTER_CHANNEL(CDR(remoteUid))=foo); might do what you want
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mikael Fredin
Sent: Tuesday, June 10, 2014 11:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
I have found here http://www.voip-info.org/wiki/view/Asterisk+RTCP that
there has been a patch for RTCP of Asterisk 1.4.
Does this mean that starting with Asterisk version 1.6 RTCP call control is
working correctly?
Kind regards
Jan Gaida
On Mon, May 12, 2014 at 2:40 PM, Jan Gaida
After reading about the 2 major SSL (and TLS?) weaknesses discovered this
year, I was wondering how it affects asterisk.
Does the SIP authentication use TLS - or something that was recently broken?
Is there a risk of exposing passwords?
Thanks!
--
On Tue, Jun 10, 2014 at 4:44 PM, Michelle Dupuis mdup...@ocg.ca wrote:
After reading about the 2 major SSL (and TLS?) weaknesses discovered
this year, I was wondering how it affects asterisk.
Asterisk uses OpenSSL for TLS. So, the answer is, it depends on the version
of OpenSSL that was
Hi,
I'm writing b/c I'm having a crash that happens a lot... it happened
without better_backtraces and then I went ahead and recompiled with it.
Then it stopped until today.
You can see the crash data at http://pastebin.com/1vbrWepr (bt, bt full
and thread apply all bt)
The extensions.conf is
On 10-06-14 23:44, Michelle Dupuis wrote:
After reading about the 2 major SSL (and TLS?) weaknesses discovered
this year, I was wondering how it affects asterisk.
Does the SIP authentication use TLS - or something that was recently
broken? Is there a risk of exposing passwords?
Asterisk'
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