ng it to MD5 in iax.conf, it is good policy anyhow not to
send plaintext passwords.
Tim Panton
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
ning and troubleshooting of
existing systems.
Can we lend a hand?
Yeah, dig into the bug tracker pick up a task and start working on
bugs and doc
for Asterisk ?
Or contribute to the discussion on this non-commercial list ?
www.sjobeck.com
s: 5
RetryTime: 300
WaitTime: 45
Context: serverdown
Extension: s
Priority: 1
which effectively 'bridges' 2 arbitrary parts of the dialplan:
exten 60 in the default context
and
exten s in the serverdown context
Hope that helps.
Tim.
Tim Panton
[EMAIL PROTECTED]
tions visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Tim Panton
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailma
Would anyone give me suggestions on how to do this? Is format_mp3
stable enough for something like this?
If you went the java applet route you could use the Tritonius
implementations of the following codecs:
ulaw, alaw and GSM. They are all GPL or LGPL.
Tim.
Tim Panton
[EMAIL PROT
it's iax.conf? Not only is
that a bit expensive
computationally, but it also allows an attacker to test 10 (say) keys
for the price of one.
Keys are for authentication not identification.
Tim.
Tim Panton
[EMAIL PROTECTED]
___
--Bandwidth and
suppression, IAX doesn't support it.
At a guess I'd say either:
a) your softclient has a bug in the audio code (it isn't sending data
every 20ms)
b) you don't have a suitable timing source on your asterisk for
meetme to use.
Many thanks
Steven Langley
Tim Pan
that show
this sort of problem, we are looking to deploy on IAX, and
a bit of warning would be great.
T.
Tim Panton
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update opt
P4 with 512MB RAM
I don't suppose you have an ethereal packet capture from a
bad call ???
Or a description of the 'badness'?
I'm doing stuff in IAX2 at the moment and might be able to spot a
problem.
Tim.
Tim Pan
major support issue from
the itsp's perspective.
There is an RFC draft, the latest version is complete enough to
implement IAX2 without
asterisk source.
I know it isn't a standard, but it is published.
Tim.
Tim Panton
[EMAIL
oes call pickup work between different protocols?
Never had a need to do pickup with iax, so don't have a clue.
As I recall, the callgroup keyword only applies to sip and zap
channels.
It doesn't work between protocols.
Tim Panton
[EMAIL PROTECTED]
_
packets.
Anyway, so I went back to a plain text file for sip.conf. What a
dissapointment.
I do think there might have been a work around available there.
Doug.
Tim Panton
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.
ing user or peer is found, the call is sent to the context
defined in the [general] section of sip.conf.
--
(I know that doesn't entirely explain the behavior but it is a
start)
I'm guessing that the sql query immediately before your extract was a
name search that c
of the
choppy
sound?
Asterisk > 1.2 have the command
iax2 show netstats
Which shows the numbers of dropped packets/jitter etc per connection
Regards,
Stojan Sljivic
Tim Panton
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided
On 15 Mar 2006, at 09:02, Stojan Sljivic - GDS wrote:
Hi,
Are you using Trunked IAX?
Currently we do not use trunking.
How many calls at a time?
All the test we have performed so far were with only one active call.
What codecs are you using?
We have set the bandwith=low, so I think that
On 14 Mar 2006, at 16:44, Stojan Sljivic - GDS wrote:
Hi,
We have two Asterisk servers connected over IAX, with very limited
bandwidth
256Kbs.
When we make calls between these two Asterisk servers the sound is very
choppy, no matter whether we use jitter buffer or not.
However, when we make
82.163.107.203 (D) 255.255.255.255 23020
Unmonitored
z212.158.206.61 (D) 255.255.255.255 56598 Unmonitored
Note the middle 2 are from the same address but have different port
numbers.
Tim
Tim Panton
[EMAIL PROTECTED]
_
act you do best to turn off any port forwarding. That way your
NAT device will allocate
different ports for each client, sharing the same IP address.
This works because IAX re-registers (or qualifies) the connection
every 60 seconds, which is enough
to keep the mapping in most NATing router&
1.2.4 and zaptel 1.2.3 (the same
CPU and load) sometimes behave in the same way. I already exclude
CPU and load to be reason of this behavior.
What technology links you to the PSTN ?
What is your clock source (zaptel card ? ztdummy?)
Tim Panton
[EMAIL
2.168.2.1:4569]
USERNAME: hayley
REFRESH : 1800
You are only allowing Hayley to come from 192.168.0.0/255.255.255.0
but the rx packet is from
192.168.2.1
Asterisk is saying 2 != 0
Tim Panton
[EMAIL PROTECTED]
___
--Bandwidth
On 4 Mar 2006, at 08:30, Paul Hewlett wrote:
On Thursday 02 March 2006 22:19, Matt Schulte wrote:
All, I'm not sure how to word this question but we're noticing a
lot of
our asterisk boxes no longer have multiple asterisk child processes.
i.e. doing a 'ps ax' reveals only 1 asterisk PID when
,crc4
span=2,2,0,ccs,hdb3,crc4
span=3,3,0,ccs,hdb3,crc4
span=4,4,0,ccs,hdb3,crc4
Assuming you have 4 PRI connections that is.
I don't know about the A104D, but I guess it is the same.
Tim Panton
[EMAIL PROTECTED]
___
--Bandwidth and Coloc
On 19 Feb 2006, at 14:54, Dumpolid Exeplish wrote:
Hi everyone,
Can anyone give me suggestions on any equipment that can connect
from VOIP to a GSM gateway (channelbank that can load up to 30 sim
cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked
at 2z's Stargate (which has
On 19 Feb 2006, at 06:04, Lee Howard wrote:J Poz wrote: Using an analog line is not an option for my service. My application runs on a ROOT SERVER of an ASP. So I can do anything I want to the server but I can't connect to or get external analog lines. So my options are doing faxing via the Interne
On 5 Feb 2006, at 21:11, Michiel van Baak wrote:On 22:38, Sun 05 Feb 06, Cosmin Prund wrote: Hello everyone. I'm again bothering you with a bit of a problem, hopefullynot really a problem. I just need someone to tell me this is ok :-)I'm planning on having two * machines on the open internet (ie: n
On 4 Feb 2006, at 23:33, Chris Bagnall wrote:You need to get BT to agree and allocate or port the numbers.You need to agree how many digits BT will pass on to you (probably 1925838395 but possibly just the last 2) I don't know the number of digits that BT pass through on a PRI, but on aset of BRIs
[EMAIL PROTECTED] wrote:
Hi,
We are ordering a bank of numbers from our provider BT. We will have an
ISDN30 with 8 channels enabled. Is it possible to do this? Is this
known as DDI? Can anyone give tips on how to configure the Asterisk
server so that users are available on the extensions.
On 2 Feb 2006, at 08:09, Cosmin Prund wrote: Brrghhh: Bandwidth calculation is really foggy for me: Using the calculator I’m getting about 23 kbps for both incoming and outgoing. What does this mean: Is a 64kbit link used at 71% capacity ((23+23):64) or is it used at only 35% (23:64)? Will this var
Brent Torrenga wrote:
Thanks for your input, everyone, but I still think it is on Teliax's end...
I will present our collective thoughts to their tech.
Kevin,
I am using IAX. When I turn on IAX debug, I get:
--SNIP CLI OUTPUT--
-- Executing Dial("SIP/Brent_ring-bcf7", "IAX2/teliax
On 24 Jan 2006, at 20:46, Andy Hamilton wrote:Has anyone seen this before:Jan 24 18:24:56 WARNING[7959] chan_iax2.c: Received mini frame before firstfull voice frame http://lists.digium.com/pipermail/asterisk-users/2005-April/103136.htmlBut is is a _warning_ It means (at the very least) that you wo
On 24 Jan 2006, at 01:42, Douglas Garstang wrote:I'm trying to think of a way to store/represent the Asterisk .conf files. One method is to store them in MySQL in some format, and then write some scripts to query MySQL and generate the conf files before doing a reload.MySQL is pretty heavy handed t
On 20 Jan 2006, at 11:49, Alessio Focardi wrote:Hi,I would like to know if there is a way to detect the status of a spanprior of sendig a call across it from the dialplan.I was asked to set up an * server with 2 spans connected to the telco and use the second asfailover for the first.I checked that
On 16 Jan 2006, at 11:05, Xavier Gil wrote:We run a asterisk 1.2.1 on a HP Proliant ML310, PIV 3Ghz 2 Mb L2 cache, 1 Gb Ram. We have a TE210Pdigium card configured for E1. This pbx has been running for almost a moth before giving this problems, we have called our telcoand seens that in their side a
On 14 Jan 2006, at 10:47, Koopmann, Jan-Peter wrote:Hi,this is the scenario:One * is placed in a central location with more than enough up/downbandwidth. One * is placed behind a DSL 3000/384. Both * are linked viaIAX trunking. Everything is fine until the upload channel of the remotesite is filled
On 6 Jan 2006, at 16:28, Joan Bautista wrote:Hi, I haven't found anything about the message below on the mailing list, Does anyones knows why this notice is being appearing? -- Executing Dial("Local/[EMAIL PROTECTED],2", "IAX2/CallOut/12365533643|30|otT") in new stack -- Called CallOut/1236553
On 4 Jan 2006, at 13:28, Francisco Pérez Botella wrote:El Miércoles, 4 de Enero de 2006 12:28, tim panton escribió: On 3 Jan 2006, at 19:10, Francisco Pérez Botella wrote: Hi.I will have to manage From asterisk to clients IP-phones, so bieflythe ideais to multiplex voip flows in large packets and
On 3 Jan 2006, at 19:10, Francisco Pérez Botella wrote:Hi.I will have to manage From asterisk to clients IP-phones, so biefly the idea is to multiplex voip flows in large packets and multicast them from asterisk/AP to client stations. flows from client stations to asterisk gateway go unicast. I won
On 21 Dec 2005, at 22:31, Matt Roth wrote:List users,I have some additional information related to the dropped audio. As always, I'd appreciate any help interpreting it.I set up an extension that calls the Milliwatt() application and is digitally recorded by the Monitor() application. Calls place
On 21 Dec 2005, at 15:18, <[EMAIL PROTECTED]> wrote:Can someone help me understand how to identify frame slips from "pri debug","pri intense debug", or any other method? I am familiar with zttest, avoiding interrupt sharing, mucking with ACPI,making sure DMA is on, etc... I have a list of changes
On 4 Dec 2005, at 21:14, chawki hammoud wrote:Hi:Sorry,but i dont know what ethereal is,and for myasterisk version the iax is good on it because i madea lot of succesful iax connections with many voipproviders like "sixtel,voipjet..."Yep, I agree, your asterisk should be fine,but this provider is s
On 4 Dec 2005, at 13:33, chawki hammoud wrote:HI:i tried to write "asterisk -rv" on console but "nosuch command" messageappears,but when i make "show version" it gives methis:Asterisk CVS-v1-0-08/22/05-18:56:48 built by[EMAIL PROTECTED] on a i686 running Linux.Weird. IAX should be fine with that ve
On 3 Dec 2005, at 21:41, chawki hammoud wrote:Hi:Thanks for your answer, i tried all possible codecsand the same result the call failed,my asteriskverison is 1.0 ,I asked callshopcompany "the voipprovider" about whats the reason of the failure of thecalls and he said he didnt know whats the problem
On 3 Dec 2005, at 20:27, chawki hammoud wrote:Hi:i made the debug and look what i get: dial [EMAIL PROTECTED] -- Executing Dial("OSS/dsp","iax2/callshopcompany/0017046872001") in new stack -- Called callshopcompany/0017046872001Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type:IAX Subcl
and goes straight to answerphone, that will always answer first.Personally for mobiles I prefer to use sms for notification and voice for office/home. Tony From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of tim panton Sent: 29 November 2005 18:37 To: Asterisk Users Mailing List - Non-Co
Channel: Local/[EMAIL PROTECTED]Callerid: 01612370660MaxRetries: 5RetryTime: 300WaitTime: 45Context: serverdownExtension: sPriority: 1On 29 Nov 2005, at 15:39, Tony Spencer wrote:I'm a bit of newbie to Asterisk so I'm not to sure.I was just given the task to try and make this work.You could be corr
On 27 Nov 2005, at 13:43, Chris Mason (Lists) wrote:Is there a good quality stable (not free) IAX2 client for MAC? I have a client wants to travel and make calls and I want to avoid the SIP blocking that is a problem for travellers.I have heard good things about http://www.loudhush.ro/ But haven't
On 25 Nov 2005, at 07:51, Julian Lyndon-Smith wrote:Thanks for your help Tim:Comments inline:tim panton wrote: On 24 Nov 2005, at 10:26, Julian Lyndon-Smith wrote: I know that's a real newbie question, but I have a problem.I keep getting frame rejects, and a D-channel bouncing up and down. BT say t
On 24 Nov 2005, at 10:26, Julian Lyndon-Smith wrote:I know that's a real newbie question, but I have a problem.I keep getting frame rejects, and a D-channel bouncing up and down. BT say that it is at my end. If I stop asterisk, stop the zaptel service and restart, things seem ok for a while.Pardon
On 20 Nov 2005, at 20:40, Nitesh Divecha wrote:
Thanks Rich,
The thing is I can not set a static CallerID for all outbound
calls. For example "Set(CallerIDnum=2021235395|a)".
I want to pass the DID numbers which are assign to customers to
outbound provider. For example if a customer with
On 21 Nov 2005, at 00:38, Luki wrote:LD_ASSUME_KERNEL 2.4.1 ... will make the kernel do old-styleprocess-perthread posix threads. I don't have this anywhere in the startup script on 2.6.12-1.1372_FC3and still have only one process in ps:Sorry, I wasn't clear, if you _do_ have LD_ASSUME_KERNEL 2.4.1
On 20 Nov 2005, at 18:58, Rich Adamson wrote:
I guess it must be a 2.6 kernel stuff since on 2.4 it only makes 1
instance but on my 2.6 boxes, it makes about 10.
I always wondered about this and never got a clear answer. On CentOS
4, Fedora 3 and RHES 4 (all 2.6 kernels) it only shows up as
On 19 Nov 2005, at 12:42, Matt Riddell wrote:
You could do it based on the new max load/max calls values
erm, I missed these, where are they - can you point me at some docs
please ?
True. But it would be nice to have one box sitting there deciding
where to
send registrations to as
On 19 Nov 2005, at 10:48, Matt Riddell wrote:
tim panton wrote:
On 19 Nov 2005, at 03:07, Matt Riddell wrote:
Marc Storck wrote:
Hello,
I would like to know if there is a way in IAX2 and SIP to tell
a client
to register at a different server.
For example:
Client tries to register
On 19 Nov 2005, at 03:07, Matt Riddell wrote:
Marc Storck wrote:
Hello,
I would like to know if there is a way in IAX2 and SIP to tell a
client
to register at a different server.
For example:
Client tries to register at server B but server B answers with
some sort
of redirect to tell t
On 18 Nov 2005, at 22:01, Piotr A. Sygula wrote:
If teliax ever wants to connect to your asterisk box, as in if they're
providing a DID for you, you will need to allow teliax through the
firewall.
If you're the one originating the connection to them, you don't
need to open
the ingress port
On 11 Nov 2005, at 12:49, Paul Davidson wrote:
**
As someone who uses and develops Notes and Asterisk on an almost
daily basis, I can tell you two things:
1. Technically, all softphones 'support' Lotus Notes- if Notes knew
how to pass them a number, they'
On 9 Nov 2005, at 19:12, Branko Samardzic wrote:
Hi,
I am trying to make following setup
PSTN --> Asterisk_1 --- IAX2 stream over Internet ->
Asterisk_2 -->
PSTN
^
|
AGI control App
Basically this is calling card solution that accepts calls from
On 3 Nov 2005, at 11:36, Chris Bagnall wrote:
I would suggest using a pair of 4-port cards. The interrupts
alone from 5 PCI cards would kill most boxes. There is also
an octo-card, but I have no personal experience of that.
Hmm... the price is something of an obstacle - given that single
BR
On 21 Oct 2005, at 06:42, Andreas Mavrides wrote: mine is 339Mine was working yesterday, but it was not supporting the GSM codec, which was supported for the free-to48-states outbound service when it was running.How is your asterisk configured? Do you permit the GSM Codec?Tim. ___
On 18 Oct 2005, at 08:05, Matthew Simpson wrote:
GoIAX, the Asterisk community's free IAX provider, is offering free
US dids now. I loaded about 175 dids in and put up a very beta
sign in page.
Fantastic, got one, thanks.
Unfortunately I had to restrict the free us/canada outbound call
On 17 Oct 2005, at 15:06, Rich Adamson wrote:
By the way, there is a reason for this. It ensures that there is
traffic (initiated by the client) often
enough to keep the 'connection' in a NATing firewall's map of ports.
This means that a
'new' call (ie incoming) message from asterisk to the cli
On 17 Oct 2005, at 01:57, Kevin P. Fleming wrote:
Ronald Wiplinger wrote:
Ok, ok,
Thanks :-)
Combining our findings now: It seems that firefly wants to
register every 1200 seconds, but iax.conf only allows 60. How can
I stop this warning message?
Asterisk has never default
On 15 Oct 2005, at 19:58, Leif Madsen wrote:
Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk
Documentation Project, in conjunction with O'Reilly Media are pleased
to announce the official release of Asterisk: The Future of Telephony
on Friday, October 14, 2005 at AstriCon 2005 in
On 10 Oct 2005, at 19:46, lenz wrote:
In data Mon, 10 Oct 2005 14:03:55 +0200, tim panton
<[EMAIL PROTECTED]> ha scritto:
Yep, I'm working on such a thing.
I have a demo version running at http://www.westhawk.co.uk/
software/faceless/CallUs.html
You don't even need t
On 8 Oct 2005, at 09:49, snacktime wrote:
I don't know, after looking at their roadmap I don't get it. It
must be the asterisk commit policies that are driving this. They
have some good ideas, but they are going about this the wrong way
if their goal is to create a successful fork of ast
On 3 Oct 2005, at 22:54, Matt Roth wrote:
List members,
It has been a while, but I once implemented a simple shared database
over NFS, so
dredging my memory produced the following:
Future Plans and Unresolved Issues
==
I wrote Windows software for anoth
On 30 Sep 2005, at 18:14, Anders Svensson wrote: Hi all! We have to setup 2 *servers. Now I am interested in possible capacity. Server 1. Should be used for getting traffic from our Telco using IAX and send it out using SIP. No transcoding, ulaw both ways. What is possible capacity on 1 server usi
On 28 Sep 2005, at 07:26, Michael Jia wrote:Hi, Scott This is Michael Jia. So far, I searched the lists and with the following email threads http://lists.digium.com/pipermail/asterisk-dev/2004-December/008295.html I don't know what is the current working status now. Maybe somone in the lists knows
On 20 Sep 2005, at 12:12, Baris Simsek wrote:Status: Provisioned, In Alarm, Down, Active Call your provider and ask them what they see. I guess they haven't enabled it yet.Tim.___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mai
On 10 Aug 2005, at 16:48, Michiel van Baak wrote:On 08:45, Wed 10 Aug 05, Jean-Michel Hiver wrote: 1) your provider is voluntarily screwing up VoIP traffic2) some idiot purposingly fills up your pipe with UDP traffic If they fill the pipe with TCP traffic, UDP will be dead aswell. Protocols don't m
On 2 Aug 2005, at 01:10, Bill Wesson wrote:Hello list,This sounds interesting. Has anyone looked at the source code of these phoneclients. I would be reluctant to download and install software that could bea trojan software.Thanks,Bill WessonIf anyone is interested I'm (slowly) developing a GPL'd J
On 27 Jul 2005, at 09:53, James Cloos wrote:
"Rob" == Rob Scott <[EMAIL PROTECTED]> writes:
Rob> For work environments where you only get HTTP or HTTPS access,
Rob> what is the feasibility of doing IAX over HTTP?
Tunnelling tcp/ip over http/(tls/)?tcp/ip is viable, tunnelling
rtp/udb/ip or
On 18 Jul 2005, at 12:06, Lee Archer wrote:
Also NTL don't drop the leading 0 on incoming numbers like BT do.
What NTL do seems to vary somewhat. I think it depends on
the switch and datafill people. This has it's plus side, you can
sometimes get _exactly_ what you want, but it does mean
ther
On 12 Jul 2005, at 15:05, Tony Mountifield wrote:In article <[EMAIL PROTECTED]>,Mark Edwards <[EMAIL PROTECTED]> wrote: I'd be interested to understand where you read about the 'sequence numbers' issue. This sounds like it might relate to the problem I am experiencing. It's either that, or the DTMF
On 12 Jul 2005, at 15:05, Tony Mountifield wrote:In article <[EMAIL PROTECTED]>,Mark Edwards <[EMAIL PROTECTED]> wrote: I'd be interested to understand where you read about the 'sequence numbers' issue. This sounds like it might relate to the problem I am experiencing. It's either that, or the DTMF
On 2 Jul 2005, at 08:48, Dinesh Nair wrote:
On 07/02/05 02:15 Matthew Boehm said the following:
according to the wiki, I should be able to do this:
exten => _9./3003,1,Set(CALLERID(number)=281443)
exten => _9./3004,n,Set(CALLERID(number)=281444)
exten => _9./3005,n,Set(CALLERID(numb
On 29 Jun 2005, at 04:51, Matthew Boehm wrote:
Hey gang,
I've been able to use sipp to produce some call volume on our
asterisk
server. The server has no problems handling 50 simul calls. But
then again,
no RTP is being done. I tried to use the rtp echo ability of sipp
but that
doesn
On 29 Jun 2005, at 04:51, Matthew Boehm wrote:
Hey gang,
I've been able to use sipp to produce some call volume on our
asterisk
server. The server has no problems handling 50 simul calls. But
then again,
no RTP is being done. I tried to use the rtp echo ability of sipp
but that
doesn't
On 28 Jun 2005, at 01:34, Eric Wieling aka ManxPower wrote:
I have no idea. But since it's NOT the same part number, I would
assume no. Perhaps a call to Digium would be in order?
Florin Mandache wrote:
As is on that page :
D/300JCT-1E1 E1 + 30 voice so is compatible ??!??
-Original
On 23 Jun 2005, at 10:48, Waldo Rubinstein wrote:
I'm trying to figure out how much call load I can put on a Dual
Xeon 2.4 Ghz Asterisk server acting strictly as an IAX2 call
director, as show in the diagram below.
The idea is that I have N number of gateway asterisk servers
connected to
On 17 May 2005, at 10:44, Peter Svensson wrote:
On Tue, 17 May 2005, tim panton wrote:
The 'if possible' thing relates to filesystem design.
Almost all of the native UNIX filesystems support mv as an atomic
action
- but only within the same filesystem.
(Imagine you create the f
On 17 May 2005, at 02:21, Corey Hickey wrote:
Hello,
The company I work for deploys and manages telecom hardware for
small- to
medium-sized businesses. My boss has asked me to investigate
Asterisk as a
possible PBX for deploying to customers along with IP phones. The
general
layout would be:
On 16 May 2005, at 22:54, Jean-Denis Girard wrote:
Andres Paglayan a écrit :
File::copy does copy, it re-writes the file,
you need to move it.
so when the the pointer is placed the file is already there.
Well from File::Copy man page, about the move() function:
"If possible, move() will simply rena
On 15 May 2005, at 17:26, Steve Underwood wrote:
Michael Welter wrote:
What is in the bug tracker helps make things clearer to people
who know what they are doing. What we need is something that
makes things clear to laymen. Saying internally and externally
clocked doesn't cut it. It nee
On 13 May 2005, at 23:38, Terje Elde wrote:
Hi all,
Sorry if this is too far off-topic, it sounds potentially
interesting to others though. I'll be brief.
Rich Adamson wrote:
I gave up (for now) trying to make spandsp work with the digium TDM
card. Instead, I signed up with www.trustfax.com at
On 11 May 2005, at 21:20, David John Walsh wrote:
being from a telecoms background, the thought of a single asterisk box
solution (even in a low production environment of say <10 phones)
worries me slightly!
Sure, and so it should, but not too much.
There is a complex trade off here, and you as you
A Physicist friend of mine named all the machines in a class C afterthe chemical elements (Hydrogen was x.x.x.1 etc).Our work systems are named after the (fictional) islands in the Earthseabooks. T. http://www.westhawk.co.uk/ ___
Asterisk-Users mailing l
On 11 May 2005, at 12:41, Matt Riddell wrote:
Anton Krall wrote:
Is there any good IAX2 or SIP free web client? Im looking for
something open
source or free preferably IAX2 for integrating into a web site...
Any leads?
Sounds like you're looking for the IAXClient libaries. There are
many ex
On 24 Apr 2005, at 18:53, Kumara Jayaweera wrote:
Hi all,
What is the best client's protocol for my softphones in Windows
pcs? and
what is the best way for connecting clients, I meant should I use
one
protocol for all the clients or some mix (SIP/IAX/oh323) of
protocols?
what
is
On 22 Apr 2005, at 16:02, Mark Phillips wrote:
Nope. This didn't work. I reordered the settings as described below
and then did "restart gracefully" at the CLI.
No change.
The received wisdom is that you need to power cycle a box if you change
zaptel.conf
it forces a cold start at _both_ ends of
On 19 Apr 2005, at 17:57, Javier Godinez wrote:
Well, we have three separate networks all routed through a sunray
session server and it would be neat if we could talk to each other
through the sunray server, depending on the network both users are on.
It's not like I came up with this architecture
On 13 Apr 2005, at 05:42, Brian Capouch wrote:
Rick from Digium got published yesterday.
http://www.nwfusion.com/news/tech/2005/041105techupdate.html?
nlt&code=nltechupdate1476
Note that a newer version of the IAX RFC is imminent.
If they are planning to go through the full RFC process, I may
be
On 8 Apr 2005, at 20:02, Bruno Hertz wrote:
"Damon Estep" <[EMAIL PROTECTED]> writes:
Why? I'd say it's only a config issue. As long as the google group
has this mailing list as it's only feed and posting to the group
is equivalent to posting to the list everything should be fine.
How do you propos
On 5 Apr 2005, at 15:33, Ugur GUNCER wrote:
Hi
Does anyone know what isthe best codec for sound syncr. And quality
with
asterisk+zyxel p200w
I found 2 that work acceptably.
If you have a good WiFi signal and are not using WEP then (a/u)law
work ok with the P2000W
If your Wifi signal is less than p
On 5 Apr 2005, at 21:09, Richard Dutton wrote:
Hi,
I've seen from the Asterisk Hardware list that the Dialogic
D/300JCT-1E1 and
D/600JCT-2E1 cards are supported by Asterisk, can anyone tell me if the
D/300SC-1E1 and D/600SC-2E1 cards are as a client has quite a few of
these
particular model and w
On 4 Apr 2005, at 09:25, Shaoul Jacobson - TELLINK wrote:
Hi,
QoS is nice (and important) but only works within a FULLY controlled
end to
end link.
Inside a BIG enterprise LAN, on leased lines its OK.
Using end to end MPLS should also be ok
Mind that some provider sell MPLS but it is not their own
On 25 Mar 2005, at 14:35, Chris Mason wrote:
I gave up on tape as being a nightmare to maintain, I now back all my
servers and workstaions using backuppc. One linux server with a 5
device
RAID can easily backup 100 workstatons and several servers beacuase of
the
pooling system used. For a smaller
Mark Charlton wrote:
Hi all
I want to send an SMS message whenever I get a voicemail left on my [EMAIL
PROTECTED]
0.6 box. I don't have any pstn attached the the box, and I am running FWD,
voipuser, and alg as providers for various routes and redundancy. I can
find a number of providers for sendi
Jim Singh wrote:
In our setup, outbound call volume frequently exceeds
the line capacity of the DSL line. We do not want to
move to another codec to better utilize the line, but
instead wish to automatically divert overflow to the
Long Distance T1 when the DSL is "full". Ideally the
system would al
On 18 Mar 2005, at 20:22, Paco Perez wrote:
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
401 - 500 of 535 matches
Mail list logo
