bviously in an error. This is surprising me :-(
How to deal with this without adding more network complexity like routing (by
putting the
phones into a subnet or other network)?
Kind regards,
oh
--
O. Hartmann
pgpWNg1HaA9AF.pgp
Description: OpenP
Am Tue, 10 Oct 2017 09:32:54 +0200
Guido Falsi <m...@madpilot.net> schrieb:
> On 10/09/2017 23:56, O. Hartmann wrote:
> > I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current.
> > Asterisk is
> > behind a NAT router, the physical setup is very much a tri
outer could have a built in sip helper that is rewriting the contact
> for your packets.
>
> On Mon, Oct 9, 2017 at 2:56 PM, O. Hartmann <ohartm...@walstatt.org> wrote:
>
> > I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current.
> > Asterisk is be
de_DE contails all the .sln16 and .gsm files, owned by asterisk:asterisk, from
the source
above. Since the prompting of numbers and the date works well, but not the
voicemail
prompt, there is something fishy I can not fathom.
For your help I'd like to thank in advance,
Oliver
--
O. Hartmann
Ic
Am Sun, 27 Aug 2017 16:47:06 +0200
Guido Falsi <m...@madpilot.net> schrieb:
> On 08/27/2017 15:53, O. Hartmann wrote:
>
> > and in voicemail.conf I have
> >
> > [general]
> > [...]
> > tz= german ; Timezone from zonemessages below.
It might sound stupid and a kind of "noobish", but I have serious trouble with
registering one of my ITSP to Asterisk 13, running on a FreeBSD 12-CURRENT box.
The following is seen in the log and anything seems somehow "normal", my PBX
tries to
REGISTER, receives 401, and then nothing
Am Sat, 2 Sep 2017 09:58:09 +0200
"O. Hartmann" <ohartm...@walstatt.org> schrieb:
Is this question to "blunt" for this forum? The background is, that I have two
ITSP
providing VoIP. One works with Asterisk 13 like a charme, but the other one
not. This
specific ITSP cl
I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is
behind a
NAT router, the physical setup is very much a trivial one. The Asterisk PBX is
supposed
to act as the telephone gateway for several VoIP/SIP phones.
I'm using throughout pjsip as configuration, I have no
an not see anything unusual with the underlying OS or some critical debug
messages from
asterisk itself.
Any ideas?
Kind regards,
O. Hartmann
[...]
==>> START [Nov 15 13:21:06] == Setting global variable 'SIPDOMAIN' to
'192.168.2.1'
[Nov 15 13:21:15] == Using SIP RTP Audio TOS bits 184
