This is from one side and from another side really I was interested to know how
to configure two peer config files? What there names and how to let asterisk
dealing with these two files?
--
>
> Message: 19
> Date: Mon, 2 Feb 2009 17:26:32 -0600
> From: Tilghman Le
2009/2/3 Ex Vito
> (my 2c, Portugal Based)
>
> - Most really small installations are PtMP (that's the "default" you
> get when ordering a BRI)
> - You also get 3 MSNs and an NT.
> - You can order a TA instead of the NT.
> - You can order PTP + optional DDIs in blocks of 10, but you need to
>
Dear all, I've implemented an Asterisk 1.4 with SIP service for voip and
video. So I can establish a voip + video connection *one-to-one*
onlyit works OK.
But I'd like to implement a videoconference *one-to-many* in order to
intercommunicate many clients, is it possible with Asterisk 1.4 ??
Hi,
Here is zaptel.conf:
# Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand
edit
# Zaptel Channels Configurations (zaptel.conf)
#
loadzone=us
defaultzone=us
#Sangoma A102 port 1 [slot:1 bus:7 span:1]
span=1,0,0,d4,ami
e&m=1-24
and /etc/asterisk/zapata.conf:
;autogenerat
> This whole thread is getting stupid and I'd hope the people involved would
> desist from this O/T drivel.
>
> If you want a switch go to the shop, hand over some money and buy one... Like
> every one else does and they're perfectly happy with their purchase.
>
> The O.P. is not going to change
On Tue, Jan 27, 2009 at 10:21 PM, Pascal Bruno wrote:
> Is it possible to retrieve the DIALSTATUS variable when placing call through
> a call file. This variable is set when using the Dial() application from
> the dialplan, but I am using a call file for my current application and need
> to get t
Software:
dahdi-linux-complete-2.1.0.3+2.1.0.2.tar.gz
asterisk-1.6.1-rc1.tar.gz
Hardware:
4-port fxs card
Example:
# /etc/init.d/dahdi status
### Span 1: WRTDM/0 "wrtdm Board 1" (MASTER)
1 FXSFXSKS (In use)
2 FXSFXSKS (In use)
3 FXSFXSKS (In use)
(my 2c, Portugal Based)
- Most really small installations are PtMP (that's the "default" you
get when ordering a BRI)
- You also get 3 MSNs and an NT.
- You can order a TA instead of the NT.
- You can order PTP + optional DDIs in blocks of 10, but you need to
be explicit.
- Larger inst
you need to port you zaptel.conf & zapata.conf (might be
channel-additional.conf in trixbox)
Bart
- Original Message -
From: "Jeff LaCoursiere"
To:
Sent: Monday, February 02, 2009 6:24 PM
Subject: [asterisk-users] RBS T1 DID issue
>
> Howdy,
>
> New installation, trying to connect a
> >Lets start with some logical points here:
> >
> >1) 24 Ports x 15.4W/Port = 369.4Watts + Switch Power = ~400Watts... now
> >Power Supply isn't that efficient so you're getting probably a 500Watt
> >Power Supply (assuming 80%)...
>
> It'd still be a 400W PSU if it supplies 400W.
>
> >2) with a 1U
Howdy,
New installation, trying to connect an RBS T1 with AMI/D4 framing and E&M
Wink. Using a Sangoma A102d and asterisk 1.4.22-2 on Centos5 (Trixbox
2.6.2.1).
Outbound calls work fine, but inbound calls fail to read the DID
information, and with debug set to 10 I get the following:
[Feb
Singer XJ Wang wrote:
>Honestly, how are you guys expecting a 24 Port POE to be fanless?
>Lets start with some logical points here:
>1) 24 Ports x 15.4W/Port = 369.4Watts + Switch Power = ~400Watts... now
>Power Supply isn't that efficient so you're getting probably a 500Watt
>Power Supply (a
Hello all,
Anyone can tell me the cause of the problem that I am experiencing
with the GTalk channel? Please advice. I need to make this channel
work.
Thank you in advance.
GNUbie
On Fri, Jan 30, 2009 at 10:29 PM, GNUbie wrote:
> Hello Grygoriy,
>
> Below is the contents of my rtp.conf file on
"Wilton Helm" wrote:
>>A modern switched mode PSU ought to be more than 90% efficient,
>In theory, yes, in practice, not likely. It is harder to get high
>efficiency from an isolated supply than a non-isolated one. I get ads
>from IC manufacturers all the time about there 90 to 95% efficient
>
Steve Underwood wrote:
>Gordon Henderson wrote:
>> On Mon, 2 Feb 2009, Steve Underwood wrote:
>>> Bernd Felsche wrote:
Ian Cowley 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 suppl
On Mon, Feb 2, 2009 at 8:39 AM, Idris AVCI wrote:
> In my situation AMI is not an option. When somebdy puts a call on hold, on
> asterisk console I can see messages like "Started music on hold, class
> 'default', on SIP/" and "Started music on hold, class 'default', on
> SIP/".
On Monday 02 February 2009 01:39:09 pm David Gibbons wrote:
> Have you tried configuring two peer config files and setting the externip
> parameter in each of them differently to your two public ips?
What's continually shocking to me are people who make suggestions on a
list when it's clear they h
On Monday 02 February 2009 12:44:05 pm bilal ghayyad wrote:
> If that code in the below link worked, will I be able to have two SIP (IP
> Trunk), both send for same destination IP:Port, but from different source
> IP's? So the destination will authorize me in my two different IP's?
Yes, that is pr
I don't think scary is a strong enough wordterrifying? horrifying?
abominable?
PaulH
Steve Totaro wrote:
> Your carrier is running Trixbox? That is scary.
>
> Anyways, they are obviously routing calls to the wrong machine. If
> your side worked properly before and now does not, then they
Thank you Mark. I did try it out myself and figured out that it did work as
I wanted. Thanks for the quick reply though.
Nicholas Blasgen
Partner / Network Operations
Refractive Dialer LLC
408.395.2110 (w)
408.497.9796 (c)
On Mon, Feb 2, 2009 at 12:06 PM, Mark Michelson wrote:
> Nicholas Blas
David @ULC schrieb:
> vicidialnow*CLI>
> -- Executing AGI("SIP/66.54.140.46-b7800468",
> "agi-VDAD_ALL_inbound.agi|CIDL
>
> OOKUPRC-LB-SALESLINE-936998-Closer-park--999-1--
>
> ---TESTCAMP") in new stack
> Feb 2 14:53:09 NOTICE[18377]: chan_local.c:526 loca
On Mon, Feb 2, 2009 at 7:53 AM, Steve Howes wrote:
>
> On 2 Feb 2009, at 09:46, Benny Amorsen wrote:
>
>> Jeff LaCoursiere writes:
>>
>>> 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.
CLI Output :
vicidialnow*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged
Nicholas Blasgen wrote:
> I'm trying to figure out how to listen in to a channel that I specify.
> I have the impression I've seen this done via Flash web controls, but
> I'm trying to write something myself and I can't figure out what command
> would be used. ChanSpy looks great, but I don't
Have you tried configuring two peer config files and setting the externip
parameter in each of them differently to your two public ips?
Dave
-Original Message-
From: bilal ghayyad [mailto:bilmar...@yahoo.com]
Sent: Monday, February 02, 2009 2:32 PM
To: 'Asterisk Users Mailing List - Non-
My provider has one IP and one port ONLY, I need to send for him the calls from
different IP's on the same Asterisk machine, how?
Regards
Bilal
--- On Mon, 2/2/09, David Gibbons wrote:
> From: David Gibbons
> Subject: RE: [asterisk-users] Sending Calls via SIP trunk from two different
> IP
Okay, point out one reasonably priced PoE switch that has it.
Christian Victor wrote:
2009/2/2 Singer XJ Wang mailto:w...@pythian.com>>
[snipped]
You can do that by using fans other than the tiny, whiney,
40mm fans
that vibrate at 6000 to 18,000 Hz. A couple of
I'm trying to figure out how to listen in to a channel that I specify. I
have the impression I've seen this done via Flash web controls, but I'm
trying to write something myself and I can't figure out what command would
be used. ChanSpy looks great, but I don't see how to specify the channel.
I
If your provider has two different IP addresses at its endpoint, you could use
iproute2 (source based routing) with two local source addresses to make sure
that there is a one-to-one mapping of source address to destination address.
Then you could have two peer definitions and an address=declara
2009/2/2 Singer XJ Wang
> [snipped]
>
>> You can do that by using fans other than the tiny, whiney, 40mm fans
>> that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin
>> fans at the back or front, pushing air in (hence the deep
>> dimensions), but the top and bottom would need recess
Hi,
I'm new to the concept of "SIP presence" and was hoping someone could lend me a
hand.
All I really need is a client application (or I could write one) that would run
on a user's computer. This app would need to show the "state" of a few SIP
extensions (something like what I would get on th
If that code in the below link worked, will I be able to have two SIP (IP
Trunk), both send for same destination IP:Port, but from different source IP's?
So the destination will authorize me in my two different IP's?
Or that code will give me a chance to send from different ports to the
destina
Thanks to everyone who has replied so far; to answer a few of the follow up
questions that have been posed:
Dave -
> Which firmware load? We had all kinds of trouble with 8.4.x, after being
> stable for a few months on 8.3.x. Going back to 8.3.x made all of the
> weirdness disappear. While we'
How are you getting these 80 or 120mm fans in a 1U chassis? Remember you
got barely 45mm to play
with at the back and front of the switch. How are you going to mount a
80mm or 120mm fan on there? Are you assuming that the units mounted
above (or below) your switch is a short 1U? You can't assu
Sounds like there's some sort of firewall in place or something else
that is preventing an ACK from being received in response to the 200 OK.
Notice that the 200 OK keeps being retransmitted.
Lincoln King-Cliby wrote:
> Hi All,
>
> I posted this a couple weeks ago with no response, I'm hopi
Which firmware load? We had all kinds of trouble with 8.4.x, after being stable
for a few months on 8.3.x. Going back to 8.3.x made all of the weirdness
disappear. While we're on the cisco note, I have script to remotely reboot the
SIP firmware load Ciscos and to provision the phones based on a
On Mon, Feb 2, 2009 at 12:39 PM, Lincoln King-Cliby
wrote:
> Hi All,
>
> I posted this a couple weeks ago with no response, I'm hoping that someone
> will see it this time around and be so kind as to offer advice for resolving
> this issue (or point me in the direction of a better place to ask)
Hi All,
I posted this a couple weeks ago with no response, I'm hoping that someone will
see it this time around and be so kind as to offer advice for resolving this
issue (or point me in the direction of a better place to ask)
"Some" (but not all) calls on one of our Asterisk boxes are being
Can someone assist me on this please?
> 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 b
We got a few of those in 1U chassis.. if you think those are quiet...
Steve Underwood wrote:
Singer XJ Wang wrote:
[snipped]
You can do that by using fans other than the tiny, whiney, 40mm fans
that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin
fans at the back or front, pushin
>A number of 1U products use large impeller fans
I've got a CPU in a 1U package with an impeller fan. It sounds like a jet
taking off! Its not quiet.
Wilton
___
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asterisk-users mai
On Mon, 2 Feb 2009, Phil Knighton wrote:
> Hello
>
> Does anyone have any experience with configuring BT (British Telecom)
> ISDN2e lines to work with Patton SmartNodes - and then Asterisk?
>
> I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e
> lines - and in turn connected to
Singer XJ Wang wrote:
> [snipped]
>> You can do that by using fans other than the tiny, whiney, 40mm fans
>> that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin
>> fans at the back or front, pushing air in (hence the deep
>> dimensions), but the top and bottom would need recesses to
Yeah. They were running a Clarent switch and that's the one that came down.
They also had\have a Coppercom switch.
The Clarent was old, though I really didn't have any problems with it. I
could never get the Coppercom to work with Asterisk (though I'm an expert at
neither) and their tech supp
[snipped]
You can do that by using fans other than the tiny, whiney, 40mm fans
that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin
fans at the back or front, pushing air in (hence the deep
dimensions), but the top and bottom would need recesses to allow
sufficient airflow when the
Honestly, how are you guys expecting a 24 Port POE to be fanless?
Lets start with some logical points here:
1) 24 Ports x 15.4W/Port = 369.4Watts + Switch Power = ~400Watts... now
Power Supply isn't that efficient so you're getting probably a 500Watt
Power Supply (assuming 80%)...
2) with a
Your carrier is running Trixbox? That is scary.
Anyways, they are obviously routing calls to the wrong machine. If
your side worked properly before and now does not, then they have to
explain why.
That would be my stance anyways.
Thanks,
Steve
On Mon, Feb 2, 2009 at 10:18 AM, Mike Hammett wr
Hello
Does anyone have any experience with configuring BT (British Telecom)
ISDN2e lines to work with Patton SmartNodes - and then Asterisk?
I have a Patton SmartNode 4638, which is now connected to 3 x ISDN2e
lines - and in turn connected to our internal LAN. I'm having huge
issues configurin
They are running Trixbox 2.6.1.10 and I'm running Asterisk 1.2.12.1.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
--
From: "Mike Hammett"
Sent: Thursday, January 29, 2009 1:47 PM
To: "Asterisk Users Mailing List - Non
>A modern switched mode PSU ought to be more than 90% efficient,
In theory, yes, in practice, not likely. It is harder to get high efficiency
from an isolated supply than a non-isolated one. I get ads from IC
manufacturers all the time about there 90 to 95% efficient solutions, but these
are
Hi!
Is it possible to configure a negative TTL (number was not found in
Dundi) for DUNDI?
regards
klaus
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asterisk-users mailing list
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h
Hello.
We need to use the Radius module to send the CDR, we need to handle this
information in another server.
We're using Radiator (http://www.open.com.au/radiator/) as a Radius
Server.
As I pointed in my first mail : This not seems to be a problem with the
Radius Server, but with the Asterisk or
Gordon Henderson wrote:
> On Mon, 2 Feb 2009, Steve Underwood wrote:
>
>
>> Bernd Felsche wrote:
>>
>>> Ian Cowley 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!
>>> Th
On 2 Feb 2009, at 09:46, Benny Amorsen wrote:
> Jeff LaCoursiere writes:
>
>> 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 pac
Hi,
We're running on a * 1.4.21 system. We run about 80 SIP Extensions, mainly
ATCOM phones (and a few Snoms - 300 and 360), and have an additional 80 IAX2
extensions to iaxmodem devices for fax2email. We are rapidly growing and
will be adding an additional PRI trunk and grow to about 150 SIP &
Paul Hales writes:
> My memory of a HP procurve (a 2626 PWR from memory) was that it was
> quite noisy - have they changed?
The 2626 is either extremely noisy or fairly noisy, depending on which
you happen to get. Luck of the draw; I haven't found a way to predict
it. The 2650 is almost always i
Jeff LaCoursiere writes:
> 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. A
On Mon, 2 Feb 2009, Steve Underwood wrote:
> Bernd Felsche wrote:
>> Ian Cowley 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.
>>
>
In my situation AMI is not an option. When somebdy puts a call on hold, on
asterisk console I can see messages like "Started music on hold, class
'default', on SIP/" and "Started music on hold, class 'default', on
SIP/". I guess the only way in my scenerio is to modify
res_music
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