Displaying 20 results from an estimated 25 matches for "backhauled".
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backhaul
2003 Aug 17
1
BudgeTone NAT issues
Just for the record and to possibly help with others who get BudgeTone
phones.
My asterisk box is behind NAT, and I use Vonage, NuFone, and
iconnecthere for my "POTS backhaul."
On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102.
The BudgeTone definitely has issues wrt the RTP stream and NATting,
although unfortunately I haven't yet been able to dig
2007 Mar 07
2
VoIP over Alvarion Wireless
Hi,
This question isn't specifically asterisk related, but perhaps someone here
can shed some light or offer some insight.
Is anyone else here running VoIP over Alvarion wireless? If yes, do you
have any suggestions for what you've done to make it "work"? It seems that
no amount of traffic shaping, checking installs for error rates, lowering
error rates, or setting
2005 May 27
3
Newbie here. Tips on setting up 100 phones w anted.
>It will be about 100 phones at about 20 locations all within
>about 4 miles of each other.
Perhaps a more pressing question might be how you are going to backhaul
Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
metres reliably, and using Ethernet repeaters every hundred metres or so
isn't practical. You will need a fiber backbone or something like that. What
2009 Jan 30
4
Packet shaping & bandwidth changes
All of a sudden tonight my web browsing and ssh performance is
terrible. I''m on a cable connection and I''m wondering if it could be
due to evening bandwidth contention or ISP throttling. If so, I
suppose tcdevices numbers are out the window. Can anything be done?
- Grant
------------------------------------------------------------------------------
This SF.net email is
2008 Oct 30
2
up to 3000 lines capacity asterisk Deployment
Hello All,
I have a request from a prospectieve client to deploy a PBX capacity
that can do up to 3000+ lines within a geographic region similar to a
campus. The client wants analog lines for extensions and maybe VoIP
for some backhaul traffic while the other traffic would be carrid via
E1 channels. The client has other proposals to buy a mid, range telco
switch from alcatel or simens but i am
2010 Jun 25
2
Big time system
We are an asterisk user... small time system 50-100 users or so.
But, we have an opportunity to get into a big time telecom activity.
It would have 2000 to 30,000 user lines per city, and we would like to have
those brought back to a central location for control and because transport
can be more economical than remote site rentals, maintenance and personnel.
We could take the local lines into
2007 Apr 08
2
intermittent choppy sound over wifi link
I am experiencing a situation where I am getting intermittent choppy audio.
Here is the network layout:
Termination provider -> IAX2 over the Internet -> 20Mb fiber connection ->
router -> Asterisk
My ATA connection goes into the router between the fiber and the Asterisk
server on another interface here is the layout from me to Asterisk:
Sipura ATA (SPA1001 running
2006 May 04
4
why a perfectly fine iax2 host becomes UNREA CHABLE?
> Is anybody on this list actually using iax2 for
> anything mission-critical?
Yes. 2K inbound / outbound calls a day to 30 remote locations, aggregated to
2 PRI's tied together with IAX2. All with IP address specified rather than
hostname. All with Asterisk 1.0.9. All with 99.9% completion rate, and it
would be 99.999% if we weren't using consumer grade DOCSIS cable modems in
the
2014 Oct 17
1
Samba 4 to replicate my samba3.6 config
We are running Arch Linux as a new sever and only has samba4 available officially
I am trying to migrate my samba 3 config to work with samba 4
I currently use samba to authenticate windows users to use our Linux shares. Then using the Unix groups setup in NIS to validate the users access to a particular share.
Here is the problem.
I can see the shares using samba 4 but it uses the
2013 Jun 22
2
SIP Trunking Mantra (Origination)
Hello Everyone,
We are currently having talks with various service providers, and
trying to determine what the best way is to interconnect in order to
have access to the PSTN network. As you know there are two ways of
doing this:
Traditional PRI: Have trunks grouped into a transport layer such as
OC3/12. With DIDs attached to the group. As you many know, this
approach would also require a POP
2005 May 27
6
Newbie here. Tips on setting up 100 phones wanted.
I'm looking at setting up Asterisk for a completely IP environment.
All intercompany calls.
I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping to bring IP to replace the old
phones. It will be about 100 phones at about 20 locations all within
about 4 miles of each
2017 May 17
2
Frauenhofer signing off on mp3, ogg stream player for Macs?
It's really pretty simple.
You can download the code and build it all you want... ...for yourself.
It cannot be distributed, sold, or used commercially in any way.
That's all.
/g.
-----Original Message-----
From: Icecast [mailto:icecast-bounces at xiph.org] On Behalf Of Robert Jeffares
Sent: Tuesday, 16 May, 2017 17:03
To: icecast at xiph.org
Subject: Re: [Icecast] Frauenhofer signing
2009 Apr 06
1
IOS Interface
Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.
Jorge Mendoza
2014 Oct 16
0
Samba4 to replicate my samba3.6 config
We are running Arch Linux as a new sever and only has samba4 available officially
I am trying to migrate my samba 3 config to work with samba 4
I currently use samba to authenticate windows users to use our Linux shares using the unix groups as the valid users.
Here is the problem.
I can see the shares using samba 4 but it uses the "Domain users" group to write to the shares and not
2011 Jan 19
2
Asterisk extension not found problem...
Hi All,
I am using Asterisk for one of my projects in OpenBTS. I am having the age
old problem of "extension not found" when try to make
a call from one registered SIP phone to other registered SIP phone (two
mobile phones connected to Asterisk via OpenBTS).
The exact error thrown on Asterisk CLI is
*"chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to
2017 May 17
0
Frauenhofer signing off on mp3, ogg stream player for Macs?
I must say this is a very confusing issue.
I generate AAC+ on an open source system which downloaded some code
during installation from the 3GPP source.
It appears the encoding software is being freely distributed, and
hardware which decodes the stream pays a licence fee at point of
manufacture.
We have such hardware. In a number of locations. One brand has AAC+
only, not AAC.
I note VLC
2005 Mar 18
5
small Local telco (wifi voip) some experiences with * ??
Hello. I would like to know if somebody did a wireles voip with Asterisk PBX.
I think to deploy a wireless for about 500 potential customers, it's a 3 km
radius maximum coverage with houses without phone lines, I work for public
places telephony small enterprises ( a common bussines in Spain) so I can get
good rates from 4 telcos and do LCR at my asterisk PBX.
Is anybody did this before
2006 Mar 17
7
gsm picocells
Is anyone in the world making gsm 'picocells' which could be connected
to an Asterisk server and allow gsm mobiles to roam to them (and
therefore become just another extension) when in the office?
Obviously lots of things to consider (it's a licensed band) which I
think was the big holdup last time I asked this question anywhere.
I know there was talk about using them on aircraft
2017 May 17
0
Frauenhofer signing off on mp3, ogg stream player for Macs?
I am not a lawyer, information in this email is no legal advice!
On 17 May 2017, at 2:09, Greg Ogonowski wrote:
> It's really pretty simple.
> You can download the code and build it all you want... ...for
> yourself.
> It cannot be distributed, sold, or used commercially in any way.
> That's all.
> /g.
That only applies to the source code. You still need to license
2009 Jul 06
3
Small site survivability
We are currently moving away from a wide-spread Cisco CallManager deployment
to Asterisk. For many of our small sites we have the routers configured for
what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP
registrar. We are converting to SIP, and from what I can tell Cisco wants a
license for each router to run SRST over SIP...
So my question to the group is: What are