Displaying 20 results from an estimated 100 matches similar to: "Airspan / Arelnet GW and Asterisk"
2006 Mar 04
0
(no subject)
Hi
Does someone have a better sql query for selecting the provider used
by LCDial application than the one proposed in the tgz ? It's far
from working well with most of price lists.
I tried to tweak it somehow with more or less success.
Regards, Michel
--
Michel Luczak
michel.luczak@notola.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2012 Mar 08
17
kernel BUG at fs/btrfs/delayed-inode.c:1466!
Hi,
this shown up today. I had to do a hard reboot as graceful hanged on sync().
------------[ cut here ]------------
kernel BUG at fs/btrfs/delayed-inode.c:1466!
invalid opcode: 0000 [#1] SMP
CPU 10
Modules linked in: btrfs zlib_deflate lzo_compress ipmi_devintf
autofs4 be2iscsi iscsi_boot_sysfs ib_iser rdma_cm ib_cm iw_cm ib_sa
ib_mad ib_addr iscsi_tcp bnx2i cnic uio ipv6 cxgb3i libcxgbi
2007 May 10
0
gw, lsrc in julian''s patches
In http://www.ssi.bg/~ja/dgd.txt I read:
--
- key "gw" for ip_route_output used to select the right route for the
gateway
- key "lsrc" for ip_route_input used to find the best unicast route
between this IP and the destination address (similar to output routing
call but still makes the checks needed for input packet).
--
Could someone please provide a couple
2006 May 26
0
2 DSL providers, 1 GW IP and Vlans
Hi all, I''m trying to put a linux GW running with this seput:
Internet -> DSL Modem -> VLAN2
\
eth2.2
Linux > Lan
eth2.3
2008 Apr 21
0
Asterisk Jingle<->SIP GW Question
Dear All
I am using gtalk features with my own XMPP server "OpenFire"
I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls
from clients registered on my XMPP server to SIP devices by calling the xmpp
accounts registered as clients on asterisk.
So far so good. So if I want to call sip:1000 I call the xmpp account that
is bound to that account in extensions.conf.
2009 Apr 28
1
Gw.exe -- keyboard:X11DRV_MapVirtualKeyEx keyboard layout
Under wine versions .19 and .20 (only 2 I've tried), I get sticking keys resulting in uncontrollable spinning (or running or whatever key gets stuck. The log output has a bunch (100+) fixmes relating to keyboard (see code).
Thanks for any help!
Code:
fixme:keyboard:X11DRV_MapVirtualKeyEx keyboard layout (nil) is not supported
fixme:keyboard:X11DRV_LoadKeyboardLayout L"00000409",
2013 Dec 02
0
Zakonczenie Kup Teraz - nr: 3698918212 NOWE IPHONE 5 64GB WHITE B.LOCKA 12M GW PL FVAT23%
An HTML attachment was scrubbed...
URL: <http://lists.alioth.debian.org/pipermail/pkg-xen-devel/attachments/20131202/686a242a/attachment.html>
2004 Apr 11
1
Config docu for SIP<->PSTN gw ?
Hi all !
Have anyone a resource / link for documentation to configure Asterisk to
act as a SIP 2 PSTN gateway (ISDN PRI) ?
Thx.
Regads,
Andreas.
--
"If you want to pray. Go to the sea."
----------------------------------------------------------------
Andreas Czerniak <cognac@amcs.net>
PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=get&search=0xEDB224EC
2005 Jul 04
1
mgcp fon behind NAT gw
Hi
I've a mgcp fon (swissvoice IP10s) behind a NAT router. Configured is
NAT for both in/out going on port 2427. Now I got the following mgcp
debug messages when i try "mgcp audit endpoint <endpoint>"
----------------------------------
from 172.16.98.57:2427
Verb: 'RSIP', Identifier: '5346', Endpoint: 'aaln/1@[192.168.2.3]',
Version: 'MGCP
2005 Aug 07
0
Using * and 3rd party GW together
2005 Nov 14
1
Problem with 827-4v and asterisk as a pstn GW
Hi,
I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as
sip-to-pstn GW. The issue is that when a call comes in from the pstn,
asterisk correctly contacts the router, which in turns send a "183
Session progress". Obviously, asterisk thinks that the telephone is not
ringing (because it expects a "180 Ringing") and we have no ringback on
the pstn side. Putting a
2006 Nov 06
0
TrixBox and MP104 FXO (AudioCodes GW)
I'm trying to connect this FXO GW without any success
1) I had to configure the " " to "allow any SIP to connect , so there will be a connection. afte that when I'm dialing I get a noise. I read on the internet that I have to change the impedance (?)
2) I could not find any HOWTO configure this combination - I read that it is a lot of pain to configure the Mp10x but than
2008 Nov 13
0
cisco voice gw / cisco call manager /asterisk for voice record, ivr
Hello!
However I'm a newbie in Asterisk/VOIP/CM I would like to make sure that this system design can work:
Cisco 2811 Voice Gateway - sip trunk1 - asterisk on linux box - s?p trunk2 - Cisco Call Manager 6.0
There is also a Siemens Hicom old pbx connected with QSIG to the Voice Gateway. I would like to record all calls with mixmon going through Asterisk.
Is it possible?
Also if there is
2011 May 12
0
About minimum requirements to install PSTN GW+SIP Client
Dear all,
Could you teach me minimum required Asterisk modules, application and etc to install PSTN GW+SIP Client functionalities in a PC as shown below? I have already downloaded astersik-1.8.3.3 and dahdi-linux-complete-2.4.1.2+2.4.1 in the PC.
Analog telephone --- PSTN GW + SIP Client --- SIP Server
********************
PC
Since available
2013 Jul 11
0
have two H323 connection: one with GK, one with other GW. is it possible?
hello all,
i have a conceptual question.
i have a h323 gateway and it is connected to a h323 gatekeeper. my
question is: can i connect my gateway to another gateway directly? i mean
can these two gateways work with each other without working with
gatekeeper? or when i have connection with a gatekeeper, all calls must be
set through it?
thanks in advance,
SAM
-------------- next part
2014 Nov 07
0
Asterisk 12 - MGCP realtime gw load
Hello there,
I'm trying use asterisk to connect against mgcp endpoints, but i would like
to use realtime mgcp gw/endpoint configuration, but i cannot find any
documentation about how can I configure realtime mgcp gw configuration.
Anyone here can guide me in order to configure it.
Thank you
Best Regards
--
Jos? Seabra
-------------- next part --------------
An HTML attachment was
2015 Mar 05
0
hangup call gw FXO
On Wednesday, March 4, 2015, ricky gutierrez <xserverlinux at gmail.com> wrote:
> I'm having some problems with a vega sangoma, if a call comes into my
> ivr and hangs up, the call continues to ring and leaves hanging the
> channel, I have to restart Asterisk and everything works Ok
>
> my sangoma is a vega 50 , 4 FXO .
>
> I tried different tone of countries and
2015 Mar 05
1
hangup call gw FXO
Looking at the pastebin, the Vega device sends a CANCEL with reason:
Reason: Q.850 ;cause=16.
Cause 16 is normal clearing and suggests that the original caller has
disconnected. I would take a look at the Vega's logs
Regards,
Steve
On Thu, 5 Mar 2015 at 11:41 ricky gutierrez <xserverlinux at gmail.com> wrote:
>
>
> On Wednesday, March 4, 2015, ricky gutierrez
2008 Sep 19
1
Rendering problems in GW
I'm trying to get Guild Wars working under Wine. I'm running on a Phenom 9950 with 4GB of RAM and a radeonHD 4850. I have wine 1.1.4 and catalyst 8.8 installed on Ubuntu Hardy (8.04).
When GW starts, it seems to render correctly, except that the blocks of graphics seem to be positioned in the wrong places, so that all the screen is jumbled up. When I take a screenshot, (through the gw
2004 Sep 29
0
two links to default gw
Hello,
shaping seems to be a hot topic at this list. I hope nobody will be
bored with my silly routing problem.
What happens if there is a route to a network via two interfaces? From
the NAG2:
,-----
|While both routes match the destination, one of the routes has a larger
|netmask than the other. [...] The larger a netmask is, the more
|specifically a target address is matched; when routing