Check out the HP ProCurve Switch 2610-24-PWR
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: February 1, 2009 6:58 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Quiet 24 port POE
Hello Everybody!
My server is attempting to connect to a SIP device, but is not
succeeding to. I checked the actual packets traveling back and forth
with ngrep and I noticed some odd packets coming in.
Here is the outgoing INVITE packet sent to the SIP device:
#
U asteriskserver:5060 -
Hello List
I am setting up a small demo site using SS7 and one of the requirement is
to be able to unhide the numbers and locate exact location of the caller
(BTS ID). Vodafone uses Nokia-Siemens switch and has confirmed that the
parameters will be sent to the us.
I just want to know how do read
by using rtcachefriends=yes it was done.
--- On Sat, 1/31/09, Pezhman Lali pezhman_l...@yahoo.com wrote:
From: Pezhman Lali pezhman_l...@yahoo.com
Subject: [asterisk-users] iax clients were unregistered after 30sec
To: asterisk-users@lists.digium.com
Date: Saturday, January 31, 2009, 7:34
I can find FANLESS 24 port PoE 10/100
That's an achievement in itself. Can you post details - I have quite a few
locations where that might be useful...
TIA.
Regards,
Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Beware PoE switches that can't handle Class 3 (15W) on all ports.
Most have fans because 24 (or 48) x 15W is hot!
IanC
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: Sunday, February 01,
2009/1/31 OCG Technical Support supp...@ocg.ca
A little off topic but
I need to put a 24 port Gig PoE switch into a small office – no computer
room / rack etc. All CAT5 terminates near the owners desk (smart huh?).
I want to put a PoE switch in place, with 24 ports and Gig speed.
Sorry, but why u r using the Radius with the CDR? Not enough to access the CDR
in the /var/log/asterisk/cdr-csv/Master.csv?
Also, what kind of Radius u r using? Any suggested link?
Regards
Bilal
Hello list.
I'm having some problems with the CDR Radius in my
Asterisk 1.4. I'm
using two
Hi All;
I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and
xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with
a source IP address xxx.xxx.xxx.yyy and another call to be sent with another
source IP address xxx.xxx.xxx.yyz, I need this
I am confused as to what you are trying to accomplish. Can you be more
specific? It seems that you are making this too complicated. You say
that the remote end is providing you two SIP trunks that will come from
the same IP address. To distinguish them simply have them authenticate
with
I have the same issue, I was just asking about that. My main SIP PRI
provider identifies me from my IP address, but I have two separate PRIs
(different rate centers) with them.
From their end I get calls to xxx.yyy.zzz.xxx and .xxy and I have no trouble
getting calls, but I can only send calls
As far as I know all POE switches are quite noisy, they need to cool the
extra power consumed by the POE and hence they will run warmer than other
switch.
I know Cisco, 3COM, are very noisy but you can try other cheaper brand like
levelone or other to see if they have fans inside the switch.
Good
I guess one would have to ask whether 1000 Gb is necessary. That's a lot of
bandwidth. It might make sense to use it for central distribution. There are
also some that have one or two 1000 Gb ports that might be appropriate for
trunking and the rest 100 Mb which is probably fast enough for
On Sunday 01 February 2009 11:32:51 bilal ghayyad wrote:
I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy
and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call
sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent
with another
OK, if I send for my provider (the destination), it will authenticate based on
the IP ONLY, this is the provider system. And once authenticated me based on
that IP, it will give me all the schema related to this account. Sometimes I
need to use another schema for some calls, I am not able until
Apologies, I tend to give the impression that I am sure changes are easy: I
was merely giving an educated guess as a programmer but not specifically as
an Asterisk or even Linux programmer. I definitely could be way off in my
evaluation of the work involved.
Regards,
Mike
-Original
Ah, that makes more sense. Asterisk binding to another IP is not the
issue, actually, and even running another instance will not do what you
need. Your problem is that the OS itself will stamp outbound packets
with the main source IP of the main interface. Asterisk could be modified
to
At the risk of seeming impolite (I really am not), why not? Isn't Asterisk
able to send packets using another interface using bindaddr? The problem,
for the two of us, is that bindaddr is Asterisk-wide, and not per-peer.
Mike
-Original Message-
From:
Could you not use some iptables to do this? I don't know the exact command
you'd need but it could work something like...
If the destination port is 5060 and destination ip is xxx then route via the
default ip (so do nothing)
If the destination port is 5061 and destination ip is xxx change the
Hi Dudes,
i searched for some time for an answer for this, i found some posting from John
Todd half a decade ago [1], was there some chance in this? Is it somehow
possible to voip from ichat to asterisk? If there's no light, is this something
that could happen with enough founding, or is
I briefly glanced at the code before responding, and it does seem that if
you specify a bind address it will use that address when responding. I
stick by my comment that the change you want is not exactly simple -
unless you are very familiar with the 18,000 line chan_sip.c :) It also
Actually I think that is a good idea. In sip.conf setup the two remote
ends on different IPs (one of which is actually bogus). Outbound NAT
based on the destination, where you change the source IP to the one
expected by the provider, and change the bogus destination to the real
one.
On Sunday 01 February 2009 14:39:11 Jeff LaCoursiere wrote:
Actually I think that is a good idea. In sip.conf setup the two remote
ends on different IPs (one of which is actually bogus). Outbound NAT
based on the destination, where you change the source IP to the one
expected by the
On Sunday 01 February 2009 15:40:29 Tilghman Lesher wrote:
On Sunday 01 February 2009 14:39:11 Jeff LaCoursiere wrote:
Actually I think that is a good idea. In sip.conf setup the two remote
ends on different IPs (one of which is actually bogus). Outbound NAT
based on the destination,
Ian Cowley i...@moffat.co.uk wrote:
Beware PoE switches that can't handle Class 3 (15W) on all ports.
Most have fans because 24 (or 48) x 15W is hot!
That's the power supplied .. which'd be at the far end of the wire.
The efficiency of the PSU plays a big part in the heat dissipation.
The push
My memory of a HP procurve (a 2626 PWR from memory) was that it was
quite noisy - have they changed?
PaulH
OCG Technical Support wrote:
Check out the HP ProCurve Switch 2610-24-PWR
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Bernd Felsche wrote:
Ian Cowley i...@moffat.co.uk wrote:
Beware PoE switches that can't handle Class 3 (15W) on all ports.
Most have fans because 24 (or 48) x 15W is hot!
That's the power supplied .. which'd be at the far end of the wire.
The efficiency of the PSU plays a big
My google search says fanless...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales
Sent: February 1, 2009 6:49 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Quiet 24 port POE gig switch
What are the chances that this can get eventually wrapped in the Asterisk
source?
If this works, I will definitely consider upgrading to 1.6 before I
originally planned to.
Regards,
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
On Sunday 01 February 2009 18:23:04 Mike wrote:
What are the chances that this can get eventually wrapped in the Asterisk
source?
Well, this will have to work, first. Second, it would have to be adapted to
work for tcp and tls, too. We'd probably put it up on reviewboard and make
sure that it
On Fri, Jan 30, 2009 at 3:23 PM, Danny Nicholas da...@debsinc.com wrote:
The dialplan AFAIK doesn't cover HOLD handling. If you can spare the
overhead, you can make a daemon to watch hints and run a script whenever the
hint for a line goes to hold and changes from hold to inuse. Just run
On Thu, Jan 29, 2009 at 6:15 PM, Vieri rentor...@yahoo.com wrote:
I'm trying to do the same in the SPA8000 units but without any luck. If
anyone is doing something similar with this device then I'd appreciate it if
you could share your relevant config options (dial pattern, etc.).
Not
Assuming you are using SIP phones and IIRC, you can hint at the
codec to be used by setting the SIP_CODEC variable in the dialplan;
before Dial()'ing, of course ! :-)
I think this is still an area where asterisk needs improvement... Dynamic
codec (re) negotiation. Anyone care to correct
I was wondering if anyone can help me with a problem we have at one of
our sites.
We have setup a Asterisk Trunk to a Avaya PBX, ie ...
Avaya - Asterisk (1.2.30) - External ISDN Network
BUT They also have a Polycom VSX 7000 that with some sort of BRI
converters that plugs into the Avaya.
2009/2/2 Steve Underwood ste...@coppice.org
Bernd Felsche wrote:
Ian Cowley i...@moffat.co.uk wrote:
Beware PoE switches that can't handle Class 3 (15W) on all ports.
Most have fans because 24 (or 48) x 15W is hot!
That's the power supplied .. which'd be at the far end of the
2009/2/2 Daniel Harper dan...@harper.net.nz
I was wondering if anyone can help me with a problem we have at one of
our sites.
We have setup a Asterisk Trunk to a Avaya PBX, ie ...
Avaya - Asterisk (1.2.30) - External ISDN Network
BUT They also have a Polycom VSX 7000 that with some
31 jan 2009 kl. 02.44 skrev Mike:
Replying to my own message. How difficult would it be to add a
bindaddr (and possibly bindport) PER PEER in SIP.conf?
How much of a bounty would I have to pay to get this done you think?
Well, if you run bindaddr=0.0.0.0 Asterisk will listen to all IP's.
Hi All,
I am working with asterisk 1.4 branch
I need to know whether EVRC codec works with asterisk version or not?
If caller and callee both has EVRC support then how the asterisk will
transmit the audio with this codecs.
I need to know the working role of asterisk with EVRC while it is running.
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