Hi,We can provide UK toll-free inbound and can recommend a co-lo provider where you'll have single-hop access to our network. Feel free to contact me off list.Kind regards,Simon
On 9/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Can anyone direct me to a colo provider in the UKwhere I can
Hi Doug,On 9/25/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local lookup to see if a number is available locally, in order to find out if the number is available on one of the other 9 servers, this peer has to query all 9
Hi Ronald,On 9/19/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I am thinking if re-invite will interfere accounting.No it won't
Please help me to figure it out:Phone A is registered at asterisk and calls a gateway. If the gatewayallows re-invite than the rtp would go directly from phone A to
Hi Ben,Yes it is but you need to remember to still include[macro-stdpbx1exten]switch = Realtime/in your extensions.confSimonOn 9/6/06,
Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello all,Another question related to Realtime.Is it possible to call macros using Realtime arch?I have a macro definition
Hi Steven,The provider's implementation will have a bigger affect than any differences within Asterisk, e.g. how they are load-balancing and whether in fact SIP is serviced by Asterisk at all. Compared like-for-like within Asterisk we find there is not a lot in it, with each having their own pros
Hi Benny,The E61 handles this just fine. With SIP as the default channel to dial and no WiFi coverage, you get a message asking if you'd like to dial by cellular. Works nicely other than a few stability issues.
SimonOn 24 Aug 2006 11:23:39 +0200, Benny Amorsen [EMAIL PROTECTED] wrote:
H ==
Hi Haspers,Makes sure you have created an 'Internet tel' profile. It doesn't appear to do anything but was vital in getting it working for me. The other settings in the how to look sensible.Simon
On 8/24/06, Haspers [EMAIL PROTECTED] wrote:
Strange,What settings do you use? I followed this
Hi Andreas,That is incorrect. It works just fine through NAT providing:- The server is proxying RTP as it has no support for STUN etc.- The NAT is the basic domestic router style, not a full blown firewall requiring port mappings
SimonOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED] wrote:
Anyone
Hi Andreas,I'm on 1.0610.04.04 19-04-06 RM-89WOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED]
wrote:Simon, That is incorrect. It works just fine through NAT providing:
- The server is proxying RTP as it has no support for STUN etc. - The NAT is the basic domestic router style, not a full blown
Hi,Yes, the proxy and registrar settings are identical in my set-up. Getting registered is the hard part but you've done that. I'd be looking at the Asterisk end of things rather than the E61 to see why that authentication issue is arising.
WOn 8/24/06, Haspers [EMAIL PROTECTED] wrote:
I've
Hi Doug,We run with SOAP on both the client/server side, i.e. conventional AGI making SOAP request to SOAP server running on Apache. I'm sure FastAGI would be quicker but wasn't around and other than the overhead of making the SOAP request is very lightweight on the Asterisk side.
SimonOn 8/22/06,
We've done this with OpenVPN and it works fine. I'd recommend that the VPN server is not on the same box as Asterisk. Stick it on a firewall/gateway box giving access to the network containing the Asterisk boxes behind it. This way the Asterisk box(es) is seeing normal unencrypted traffic and the
Hi Curt,That probably suggests that with SIP they're handing off the RTP to their upstream provider and just dealing with the signalling which is very low overhead. With IAX they have to transport both unless they're interconnecting upstream by IAX and can transfer. In my experience the load is
and Realtime, I'd be grateful for any pointers.Many thanks,SimonOn 7/17/06, Simon Woodhead
[EMAIL PROTECTED] wrote:I'm expecting regcontext to create a context of regcontext and an priority 1 extension for either the value of
regexten or the peer name. The context is created, the extension says
Have a google for 'interface bonding'. You bond your two cards together to appear as a single one and then bind Asterisk to an IP address on it. The cards work in loadbalance or failover mode as you specify.
On 7/20/06, shadowym [EMAIL PROTECTED] wrote:
Has anyone had any success creating a
Thanks for the reply Brad.The relevant section of sip.conf was posted:[general]regcontext=sipregistrationIf you mean extensions.conf, I wasn't creating the extension in there other than for testing. RegContext correctly creates the context on registration but does not create the extension. If I
I'm expecting regcontext to create a context of regcontext and an priority 1 extension for either the value of regexten or the peer name. The context is created, the extension says it is created but isn't. It works fine with a staticaly defined extension of the same name as defined for regexten or
Hi folks,
I've been having a go at getting DUNDI working this evening to enable
users to register to any Asterisk box and to look them up from another.
The DUNDI part works just great (very impressed), as does the subsequent
joining of calls between the two servers but I'm struggling with
Yep, cmd setCDRUserField will do this for you assuming you have the field set up. I'd be keen to hear if anyone has a way of achieving the same thing across multiple user fields to save having to explode multiple values out of a single user field seperately.
SimonOn 6/20/06, trixter aka Bret
That sounds fine except where registrations are involved although I'd suggest you look into SRV as well as RR for the DNS to more finely balance the load for clients which support it. Doug's mail says it all where registrations are involved - not all state information is stored in the database so
We use Unison Doug and it works just fine. It isn't perfect in theory but we've had no issues in practice. Your concerns over sacalbility are resolved by implementation - do you need it on every single Asterisk box, or maybe local to just two with routing to them and failover in the dial-plan?
Hi Doug,We use Realtime SIP via a central MySQL database (2 actually in Master Master config) but registration is only available on the box to which the client has registered. Clients can register with any database and the table does get updated with some registration information (ip address,
Bad day Damon? I think your comments are a little harsh towards someone who is an active and informed contributor to the list. Jean-Michel could have ignored you but he chose to share what he could. Maybe someone else will have the complete answer to your question.
On 1/26/06, Damon Estep [EMAIL
We will Scott.
http://www.esms.com or drop me a mail off-list.
Kind regards,
SimonOn 1/24/06, scott [EMAIL PROTECTED] wrote:
HiDoes anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account.www.SipGate.co.uk
are great but they only allow 1
We've been trying Unison (http://www.cis.upenn.edu/~bcpierce/unison/) on a 1 minute cron job. There are some theoretical issues but it has been great so far. We use it to synch prompts as well as messages.
SimonOn 12/27/05, BILL GITONGA [EMAIL PROTECTED] wrote:
What is the best method of storing
Hi Bails,
We'll help. Drop me a mail off-list.
Simon
http://www.esms.comOn 12/8/05, bails [EMAIL PROTECTED] wrote:
Hi all, just got an iaxy box for a customer and its great, but!I really dont want to host and bill this customer myself and i cannot
find a voip to pstn breakout that will let him
Our free UK numbers can forward to IAX:
http://www.esms.com/services_numbers_pure_free.php
Simon
On 11/3/05, Gabor Horvath [EMAIL PROTECTED] wrote:
Dear Asterisk users,
can you suggest me a free service where I can test my IAX trunks? Thank you.
Gabor
Fax over VoIP is just not reliable in my opinion. I'd run with doing it
directly to PSTN as the other poster suggested or via Hylafax. We've
used Hylafax behind Asterisk very succesfully in the past.
SimonOn 10/29/05, KARIM MOUSLI [EMAIL PROTECTED] wrote:
my problem is to triger the transfer to
Hi folks,
Is anyone aware of a way to prevent transcoding or better still apply
some kind of weighting to codec selection based on other channels in the
call? Let's say we support g729 and gsm, a peer supports both and a
client supports one of them. We're seeing calls frequently coming in on
Hi Rene,
Yes, I've seen that but our version from CVS is a month or so old os it
may well have been rectified now. On our version reloads cause the
process to die about 50% of the time, work fine about 45% and cause it
to hang in the way your describe probably 5%.
Simon
On 19/10/05, René Enskat
Hi folks,
I've just been reading about the above command and wonder if anyone has
made use of it for load-balancing or if doing so would be completely
inappropriate!?
I'm thinking of the scenario where there are a number of Asterisk
gateways and incoming SIP traffic. From what I've read,
idle. If no agents answer it should try
something else or have the option to try something else.
bkw
On Sat, 9 Aug 2003, Simon Woodhead wrote:
Hiya,
First off, thanks to everyone involved in app_queue. Its a great addition
to
an already great system.
For my two penneth (or cents!), I
Hi folks,
We're getting some IVR/busy messages recorded for Asterisk which I
understand have to be in GSM format but we've been given a list of options
which don't mean a lot:
RAW GSM 6.10 audio stream
RAW 'byte aligned' GSM 6.10audio stream
US ROBOTICS VOICE MODEM W.O. HEADER
US ROBOTICS VOICE
Hiya,
First off, thanks to everyone involved in app_queue. Its a great addition to
an already great system.
For my two penneth (or cents!), I think the following would be good:
- the fallback method should be optional if at all possible, so it can be
set up for, say, fewest calls with ring all
Hi Uriel,
Forgive me if you've already done this, but have you checked disk space on
the mailserver? Its caught me before and might save you hours debugging
something that isn't broke.
W
- Original Message -
From: Uriel Carrasquilla [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
Hi Dave,
I think it will. We have a very similar requirement...
My aim is simply to have incoming calls identified (using CID) and logged,
then initiating a request - from the relevant client pc - to the webserver
for a specific page (the customer history/details page).
We're working towards
Hi Nick,
You'll probably run into quality problems making calls over the ISDN from
Xten via *. We did which led us to try several other softphones which were
better and worse, e.g. Pingtel was great from a quality point of view but
the interface wasn't.
We're using snom 100s at the moment which
Hey Dan,
Get a http://www.mini-itx.com/ and disguise it as a fruit bowl. She'll never
know! I can't wait until someone builds one looking like a shoe or a handbag
and then I can have them all over the house and the more I have, the happier
the other half will be!!
W
- Original Message -
You would still use PRI if you need bulk lines. You could use
channelized T1, but you get a lot more options with PRI. Currently our
phone server is in our colo rack and our phone lines are sent down to us
via our data T1 line.
Thanks Steven. I'll go investiagte that.
Cheers,
Simon
Hi Steven,
BTW, my problems where on our private T1 line that sees round trips in
the 4ms range. Our semi educated guess was that we had a problem with
the jitter buffer causing echo cancel to go nutty when our ping times
would occasionally jump to 20ms. When I turned off the jitter buffer,
the
Hylafax Dan. It isn't that elegant though as you'll need to wire an analogue
port to each fax/modem. AFAIK there isn't a virtual fax/modem provided by *
that another programme can use.
W
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, July 05, 2003
box
Hi,
What I need is a pure software solution, to avoid any other hardware to get
that functionality.
Thanks,
Dan
- Original Message -
From: Simon Woodhead [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 07, 2003 11:50 AM
Subject: Re: [Asterisk-Users] Virtual fax
Your uplink is pretty limited at both ends. I'd be
using g.729 over IAX in your situation giving enough uplink for several calls,
or a call and normal use at least. GSM is a bit too hungry for that kind of
connection.
- Original Message -
From:
Jay
Tyndall
To: [EMAIL
The snom phones (and I assume others) allow you to have multiple SIP
accounts on a single phone. The user logs in to the phone which logs in *.
The downside is that you can only log in to accounts set up on the phone
rather than any account set up on * but is useful for shared desks etc..
W
For the Snom100 is a IAX Image available at asterisk's ftp site.
Can you tell me more? Is it a patch to enable IAX or replacement firmware?
Many thanks,
Simon
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hiya,
Yes it does.
The only thing to be careful of, as we learnt to our mistake, was that a
single purchase gives you a single key for all and thus you cannot buy 10
licenses intending to use some on one server and some on another. I guess
this would be possible by special request though.
Simon
No, you can reinstall up to 3 times I believe.
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 16, 2003 2:11 PM
Subject: Re: [Asterisk-Users] G.729 Licencing..
What if you change the hardware?
The licenses are lost?
Thanks,
Dan
-
interface and for this purpose I need a system
with
a PCI bus.
I can try the codec now on this installation (notebook) and then move it
to
the new system when it will be available and still keep working?
Thanks,
Dan
- Original Message -
From: Simon Woodhead [EMAIL PROTECTED
Hey Jim,
All sounds good.
We tried a satellite system here a few months ago but couldn't get on with
it. Glad you've had more success. In theory, it shouldn't matter whether the
TCP/IP link between your sites is going over satellite, modem or any other
medium but the issues we found with
That's about right, inlcuding postage.
- Original Message -
From: nathan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 12:44 PM
Subject: [Asterisk-Users] What is the going rate for the Snom 100 in the UK?
Hi All,
What is the going rate for the Snom 100 in the
Hi Richard,
So is a PBX style the equivalent of a straightforward adapter or is there
more to it than that?
I noticed last night that our analogue phones actually have RJ11 sockets
that the BTRJ11 lead plugs into so I hooked one of them up with an
RJ11RJ11 modem lead but got nothing. Am I doing
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