Hello,
worst aspect is that - if SIP clients do not have such a timeout, and in
that case if killing an asterisk and to start it up again -
so it is nothing to do with this asterisk timeout.
Regards,
On 23 February 2010 08:44, Olle E. Johansson wrote:
>
> 23 feb 2010 kl. 01.47 skrev Kirill 'Big
Hi,
sorry if the question seems silly, but for some reason, all these phones,
modems, ATAs use TR-069 instead of SNMP ?
I have experienced it, but as user, for a small time.
Regards
On 16 February 2010 09:54, Olivier wrote:
> Hi,
>
> Phone vendors (Snom, Thomson-Technicolor, ...) are on the wa
see the DTMF method on both phones.
2009/11/14 Ignacio
> Ok, thank you very much. I will try to find any information in
> asterisk documentation about RTP.
>
> On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III
> wrote:
> > On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote:
> >> I have just e
In wireshark or ethereal:
filter -> sip || rtp
Regards,
2009/8/3 Timothy Weidner
> To make your life a little easier, you can use the following filter:
> sip or sdp or rtp
>
> Just insert that into the filter query field in wireshark and it'll show
> you what you need.
>
> On Sun, Aug 2, 2009 a
Hello Cary,
Can I see your configuration files in any form, I mean only part of BLF
settings
2009/3/25 Cary Fitch
> We are not having a problem in the BLF area, and we do qualify our remote
> phones. I don't know if "qualify" does any thing beyond pinging the
> address, but perhaps it does car
maybe it is not GS,we have the same problem with snoms 360
2009/3/24 Cary Fitch
>
> -Original Message-
> boun...@lists.digium.com] On Behalf Of Rob Hillis
> Yes. Grandstreams suck.
> [Cary Fitch] We are not entitled to your opinion.
> >
> [Cary Fitch] On a sm
;
> --
> Regards,
> Robert Broyles
>
>
>
>
> Leonja Cerebro wrote:
>
> Hi,
> Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk of
> Asterisk B (registered in Asterisk A as extension)
> to incoming call across another trunk of Asterisk B t
Hi,
Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk of
Asterisk B (registered in Asterisk A as extension)
to incoming call across another trunk of Asterisk B to extension of Asterisk
C
What the dial plan should be?
Thanks
--
We never did too much talking anyway
So don't thi
Hello All,I'm very new in asterisk.Please help - how I can write conf files
(or some example) for to delete one ext. and to add another, it means for
example:
I need to call from one asterisk to another by trunk to trunk and my dialing
(for ex.) 100#...@1.2.1.2
when the the trunk of first asterisk
Hello All,I'm very new in asterisk.Please help - how I can write conf files
(or some example) for to delete one ext. and to add another, it means for
example:
I need to call from one asterisk to another by trunk to trunk and my dialing
(for ex.) 100#...@1.2.1.2
when the the trunk of first asterisk
Thanks for asking,
This is caused by kill -9
and after starting by both ways... I cannot start DNS resolving of trunks.
Regards
2008/12/18 Philipp Kempgen
> Leonja Cerebro schrieb:
>
> > I have problem after killall -9 asterisk
> > and asterisk -f
>
> Can you narro
Hello,
I have problem after killall -9 asterisk
and asterisk -f
Asterisk stops to send to DNS resolving of trunks
Regards
--
We never did too much talking anyway
So don't think twice, it's all right
___
-- Bandwidth and Colocation Provided by http://ww
Hello,
I have problem after killall -9 asterisk
and asterisk -f
Asterisk stops to send to DNS resolving of trunks
Regards
--
We never did too much talking anyway
So don't think twice, it's all right
___
-- Bandwidth and Colocation Provided by http://www
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