On 06/11/2020 14:28, basti wrote:
> Hello,
> i try to connect my SIP Client (linphone) via VPN to FreePBX.
> The routing looks OK. I can ping the Endpoints and traffic is routing.
> I can also Register my Sip Client.
>
> debpbx*CLI> pjsip list contacts
>
> Contact:
>
>
In some circumstances I am transferring incoming calls to an external
number (cell phone). Whenever this happens at the end of the call I get
a single CDR representing the outgoing leg. There is no CDR for the
incoming leg and no trace of incoming caller id in the CDR for outgoing
leg.
Is this
Chandrakant Solanki wrote:
Hi
r u forwarding call using Originate action..
Which version of asterisk u used.
Hi
asterisk 1.6.2.0
I'm using freepbx, but I looked into the generated files: if I read it
correctly it ends up using Dial cmd.
thanks,
John
Tilghman Lesher wrote:
On Saturday 09 May 2009 03:42:04 John Fawcett wrote:
I'm on asterisk 1.6.1.0 and asterisk addons 1.6.1.0 (also using freepbx
2.5 with cidlookup module from mysql database).
There are some incompatible changes in asterisk 1.6 about MYSQL addon
application syntax
I'm on asterisk 1.6.1.0 and asterisk addons 1.6.1.0 (also using freepbx
2.5 with cidlookup module from mysql database).
There are some incompatible changes in asterisk 1.6 about MYSQL addon
application syntax for querying a mysql database.
It seems that escaping of space and single quotes is no
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Joseph wrote:
| My phone rings once and stops before playing message; how to stop
this behavior.
|
| Could it have something to do with this error:
|
| channel_find_locked: Avoided initial deadlock for '0x81c04d0', 9
retries!
|
|
| Here is the dial
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Tilghman Lesher wrote:
gcc 4.2 has a bad optimization that has yet to be tracked down, which causes
the transcoding code to be incorrectly built. There are two workarounds for
this problem: either use gcc 4.1 or enable the DONT_OPTIMIZE
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I've have installed a new Asterisk 1.4.15 system after having previously
used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though
the newer one is actually a slower processor.
On the new system, playback of gsm files is noticeably
Derek Conniffe wrote:
Hi again Jörg,
The install_misdn Makefile doesn't seem to like my SMP yes in my
kernel .config (it does a grep on the .config file, finds the line,
and tells me) - I'm going to try the chan_misdn driver anyway and the
server is an old HP netserver e800 which is a dual
Jon Dean wrote:
A plea to all!
Has anyone had any success with two or more avm fritz pci cards with either
misdn, chan_misdn, or chan_capi, and any version of linux 2.6.x?
I have managed to get misdn to load under 2.6.13 and detect two cards using
misdn-capi and chan-capi (using
Konrads Smelkovs wrote:
Isn't billion a HFC PCI card? see lspci output, if so, use bristuff
from junghanns.net
http://www.junghanns.net/en/download.html , i suggest CVS version
thanks, I hadn't thought of using bristuff.
I just followed the indications on the site where I got the Billion
Dan Austin wrote:
3. Should I consider the current features a release point, or
is there
something I missed that should be added before
packaging it?
I appreciate the feedback I have received since the last announcement,
and
apologize that I allowed work to get
Sean Rima wrote:
Using the install instructions for [EMAIL PROTECTED], I setup a FWD account,
this I
tested using X-Lite and it works okay,
Nowever I cannot make calls to fwd using Asterisk, my log showes:
Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected:
Registration Refused
I've now setup SIP for:
- internal softphones
- registering with external providers (like FWD) for making calls
- receiving calls from theese providers
For the latter step, it was necessary to forward ports from my NAT
to the asterisk server: 5060 + range of ports mentioned in rtp.conf.
I was
I've been playing around with asterisk for the past few days. One thing
which came to mind
which could be a useful addition to the meetme functionality would be
the possibility to
specify a moderator password in meetme.conf. (A moderator in the sense
that music
is heard until the moderator
Michiel van Baak wrote:
On 15:14, Sat 13 Aug 05, John Fawcett wrote:
I have successfully configured asterisk to make outgoing calls over FWD,
but cannot receive incoming calls. The console shows no messages,
even though an XTEN client on the same network has no problems receiving
incoming
I have successfully configured asterisk to make outgoing calls over FWD,
but cannot receive incoming calls. The console shows no messages,
even though an XTEN client on the same network has no problems receiving
incoming calls.
This is the relevant part of sip.conf
[general]
.
register =
I'm looking at experimenting with asterisk with an ISDN BRI and ISDN
phones (since I have these already).
I saw that the Billion card was cheap and could be used in either TE or
NT modes.
I have the following question which I couldn't answer by reading through
the manual. Maybe someone has
Christian Victor wrote:
The cards feeds no power to the s0. If the phone has its own power
supply normally it will work without external power supply. Otherwise
you will need a network terminator (NTBA) or a selfmade power injector.
Christian
Christoph Eicke wrote:
I don't know the exact
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