Displaying 20 results from an estimated 300 matches similar to: "BOTH UAs behind same FW/NAT"
2004 Jan 06
2
URGENT - micronet & asterisk on h323
hello,
my situation is
h323gw - gatekeeper - asterisk - SIP client
my problem is, that I can't make call from h323gw, when this GW is
Micronet (sp5004). A
----------- CUT -----------
-- Executing Wait("H323/ip$62.152.225.18:52434/20702", "1") in new stack
== Spawn extension (postel, 169, 1) exited non-zero on
'H323/ip$62.152.225.18:52434/20702'
-----------
2003 Oct 29
2
Call transfering, conferencing
hello,
my questns are about few * functionality.
1) how can I make call tranfer. Not call parking.
If I'm talking with some one a I want to tramnfer call to the another
extension, to the other person.
2) how can I make call confernece. Not Meetme
If I'm talking with some one and I want to join another person to our talk .
I haven't found this in any manual :(
hudecof
--
mail:
2003 Nov 06
0
which codec will be used ?
hello,
my situations is as follows.
In our comapny we are planing to have *. I'm testing it now.
If we will buy G729 codec for * ...
UA(SIP) <-> FW <-> (SIP)*(H323) <-> (H323)GATEKEEPER(H323) <->
(H323)AS5300 <-> world
the following equipment speeks G729: *, GK ,AS5300.
All call from UA to another endpoing go through *, because of mixed SIP
and H323
2003 Sep 11
1
newbie - sip, pxb, ata, nat
hi all,
I don't know how to setup asterix to work as PBX.
If I want just basic configuration with 2 SIP phones (snom, ata), what
all I have to write in the configuration files, or respectively in the
configuration of ata and snom ?
If there is any good documention available, send me URL too.
All (ata, snom) are behind firewall (nat) and astrix is on the public
IP, but I can move for
2006 Feb 22
1
TFTP server for GrandStream BT phones / need testing
hi,
I known, that this is not * related, but a lot of members of this ML
uses GS BT phones.
I have patched the atftp serber to recognize the TFTP OPTION, whis these
phone send during boot.
Patch includes
- another locations for configs, firmware and ring tones
- different FW versions for phones
- custom ring tones for the phones
You can find patch, source/unpatched/ and DEB for
2015 Feb 18
2
Master - Slave Split DNS
On Feb 18, 2015 7:43 PM, "Tris Hoar" <trishoar at bgfl.org> wrote:
>
> On 18/02/2015 07:17, aditya hilman wrote:
>>
>> Hi folks,
>>
>> I've already configured split DNS for internal-view and external-view.
Also
>> already configured the master - slave dns.
>> But i've problem with external-view zone transfer.
>> Based on the
2016 Jul 22
1
authentication problem after upgrade to Debian Jessie
Hi,
I upgraded our servers from Wheezy to Jessie. I use samba in classic mode,
with openldap backend. After the upgrade, on the PDC (srv3) everything
seems to be ok, it authetnicates, the netlogon share is accessible on it,
but on the BDC (srv7), what is the file server, the authentication doesn't
work, shares are inaccessible.
I compared and syncronized the configuration files to as similar
2015 Feb 18
6
Master - Slave Split DNS
Hi folks,
I've already configured split DNS for internal-view and external-view. Also
already configured the master - slave dns.
But i've problem with external-view zone transfer.
Based on the logs, the master notify to slave using the public ip, which is
not accessible by master to transfering the zone over public ip.
Is it possible to transfer zone over local ip for external-view ?
2003 May 03
1
Failed calls from SNOM100 to *
Hi all,
I've just been setting up such a phone with * and I'm facing some problems.
At this time there is no customized extensions & others, so I use extension
8500 (voicemail) in default config.
What I get when this extension is called from the SNOM is pasted below. The
phone ignores *'s invites and * sends a "BYE" to it. Please note the IP in
the "Contact"
2003 Jun 11
1
SIP phone behind NAT
Hi all,
--------
I have a Asterisk at a public Network (official IP address). In the local
network I have isntalled a Snom 200 IP phone and in my home network (behind
NAT) a Snom 100 device. I can dial the Snom200 device from my home location
without any problems but the Snom200 can not dial me. It always gets a "we do
not rely". I tried to forward the SIP Port (5060) UDP via UPnP
2003 Oct 06
1
Snom100 H.323 sample config
I'm trying to get a Snom100 configured with H.323. Right now, the
phone is not even connecting to the Asterisk server, so there's
obviously a problem with the snom config. Does anybody have a
sample working configuration with the snom phone, using H.323?
I've checked the archives, but everybody seems to be using SIP with
the Snom phone, not H.323.
-Tilghman
2015 Feb 18
0
Master - Slave Split DNS
On 18/02/2015 07:17, aditya hilman wrote:
> Hi folks,
>
> I've already configured split DNS for internal-view and external-view. Also
> already configured the master - slave dns.
> But i've problem with external-view zone transfer.
> Based on the logs, the master notify to slave using the public ip, which is
> not accessible by master to transfering the zone over
2015 Feb 20
0
Master - Slave Split DNS
On Wed, Feb 18, 2015 at 7:47 PM, aditya hilman <aditya.hilman at gmail.com>
wrote:
>
> On Feb 18, 2015 7:43 PM, "Tris Hoar" <trishoar at bgfl.org> wrote:
> >
> > On 18/02/2015 07:17, aditya hilman wrote:
> >>
> >> Hi folks,
> >>
> >> I've already configured split DNS for internal-view and external-view.
> Also
>
2003 May 18
3
SNOM100 GSM again
OK I did some researches and tests with it, and finally:
I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console:
WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames
My conclusion is that the snom100 utilizes MSGSM
2003 May 16
5
Snom100 GSM
Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality.
Will be any change in this subject?
THX
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2004 Jan 09
2
help
Hello freebsd-stable,
i know this is may be the wrong place to ask these question,
i just cvsup my 5.1 box today,
and during the make installworld it's stop with these following
error messages:
PASS: Test notdef detected no regression.
Running test modifiers
PASS: Test modifiers detected no regression.
Running test funny_targets
PASS: Test funny_targets detected no regression.
2003 Oct 11
1
SIP / IAX over satellite
Hi all,
------
I tried to use * over satellite, but all my effort did not succeed.
The Asterisk is behind the VSAT and is resposibel for alle the SIP
clients in a field location.
The clients are notebooks and PDA's running SJPhoen for Windows and
PocketPC. Unfortunately
I could not find any Linux Client wich worked satisfying. SJ LAbs
promised a Linux Version at the end of
August but they
2003 Mar 01
1
cannot disconnect by callee at first in SIP case
sorry, this problem is fixed by myself.
we must need set 'canreinvite=no' each user.
---
I'm try to discconect a call with SIP.
when caller make a call, 'show channels' result is following.
mack*CLI> show channels
Channel (Context Extension Pri ) State Appl. Data
SIP/mack-1bfc (default 1 ) Ringing AppDial (Outgoing
2023 Nov 01
1
Again kea DHCP-Server
Well? I?m not convinced KEA can?t be simple DHCP, though I understand that one
would never know that from reading the KEA documentation.
The following is a complete KEA configuration suitable to do a single subnet. In fact, you could probably get away with slightly less.
I?ve replaced the unique addresses from my environment with placeholders (e.g. <network/mask>), but otherwise, this is a
2015 Mar 07
4
Something like apt-cacher for CentOS/RHEL?
Hi,
For some time I've fiddled with Debian and Ubuntu LTS. There's one
really nice feature for local networks: apt-cacher, a package proxy for
APT.
My company is in the remote South French countryside, and more often
than not, schools and public libraries only have some very limited
Internet access with relatively low bandwidth, which can make the
updating process very tedious. A