On 7 January 2014 22:55, Adam Moffett adamli...@plexicomm.net wrote:
I'm asking about this scenario:
Asterisk(public IP) -- Internet -- Router (public IP) -- SIP client
(private IP and NAT)
What settings in sip.conf will give this the best fighting chance of
working?
We already have
Le 18/12/2013 11:51, Administrator TOOTAI a écrit :
Hi all,
I face a strange problem. I'm in France using Completel as operator
for the E1 line. I move a client from ccm to Asterisk keeping the 2811
gateway.
Set up is complete, outgoing and incoming calls are sended to the
2811. The
I have a multi tenant asterisk box where on tenant is receiving calls from the
caller ID as1as and they cannot pickup the call.
The caller ID also does not show up in the call log.
Thoughts?
Thanks,
--Eric
--
_
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Hello,
I see the strange behaviour that outgoing calls end after 15 minutes.
I didn't knew there is some kind of call duration limit that can be set ?
Is there ?
Using Asterisk 1.8.12.2
Kind regards,
Jonas.
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Try setting canreinvite yes on that trunk it worked on trunks I had
Some providers send a reinvite after 15 min and if Asterisk doesn't respond
then it disconnects the call something like that
-Original Message-
From: Jonas Kellens jonas.kell...@telenet.be
Sender:
On 1/8/2014 4:17 AM, Ishfaq Malik wrote:
On 7 January 2014 22:55, Adam Moffett adamli...@plexicomm.net
mailto:adamli...@plexicomm.net wrote:
I'm asking about this scenario:
Asterisk(public IP) -- Internet -- Router (public IP) -- SIP
client (private IP and NAT)
What
Am 08.01.2014 16:07, schrieb Jonas Kellens:
Hello,
I see the strange behaviour that outgoing calls end after 15 minutes.
I didn't knew there is some kind of call duration limit that can be set ?
Is there ?
Look at session-timers in sip.conf. I had to set it to refuse for a
specific
On 08-01-14 16:47, Markus wrote:
Am 08.01.2014 16:07, schrieb Jonas Kellens:
Hello,
I see the strange behaviour that outgoing calls end after 15 minutes.
I didn't knew there is some kind of call duration limit that can be
set ?
Is there ?
Look at session-timers in sip.conf. I had to set
Hi, all
Sorry for null subject last mail.
I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded
from asterisk.org). We named it Asterisk11.
I want to generate a call file to /var/spool/asterisk/outgoing. This call
will dial out to Local Channel and return to some Extens.
Then
logs ?
full log containing the call?
On 8 January 2014 14:56, Eherr email.eherr9...@gmail.com wrote:
I have a multi tenant asterisk box where on tenant is receiving calls from
the caller ID as1as and they cannot pickup the call.
The caller ID also does not show up in the call log.
Richard - Perfect, Thanks for the pointer!Regards,Fred
Original Message
Subject: Re: [asterisk-users] Asterisk 1.6.2.x Keeping NAT Alive
From: Richard Mudgett rmudg...@digium.com
Date: Tue, January 07, 2014 4:42 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
Does not show up in the cdr log.
I am going to enable an asterisk cli dump tonight and try to catch it.
I am thinking its a straight sip attack or IP attach on the sip client vs a
real call or problem with asterisk.
It's also a polycom IP 335
Thanks,
--Eric
Sent from my phone.
On Jan 8,
not sure about dial, but I Set(__var=value); and in each piece of dialplan
I set CDR(var=value);
On 31 December 2013 00:00, Igor Katson igor.kat...@gmail.com wrote:
Hi,
when one does Set(CDR(var)=value) in dialplan, the value is only set for
one record in the cdr table, but not the linked
Hello,
I recently purchased the Cepstral 6 text-to-speech engine (swift), and
am now wondering if I should have bought something else. I would like
to use Cepstral text to speech like some people use the Festival() or
Flite() applications. For example, when I do a core show application
i noticed in asterisk 10.12.3, i get messages like this:
[2014-01-08 19:03:59] NOTICE[2987]: chan_sip.c:23900 handle_request_invite:
Failed to authenticate device 305sip:3...@my.server.ip;tag=0d516e63
but not mentioning attacker ip (to be used for fail2ban)
is this expected?
--
Thanks, Tiago! By the way that's exactly the workaround I came to myself.
On Wed, Jan 8, 2014 at 2:35 PM, Tiago Geada tiago.ge...@gmail.com wrote:
not sure about dial, but I Set(__var=value); and in each piece of dialplan
I set CDR(var=value);
On 31 December 2013 00:00, Igor Katson
See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Framework
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Al lists
Sent: Wednesday, January 08, 2014 10:11 PM
To: Asterisk Users Mailing List
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