On Oct 28, 2008, at 5:13 PM, Kev Szaszvari wrote:
Hi there
Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have
* Central Management for all the phones (We dont mind if we have to
buy the software to manage them)
* Programable shortcut buttons, So i can program in on
2008/10/28 Robert Boardman [EMAIL PROTECTED]
Olivier wrote:
2008/10/5 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Kevin P. Fleming wrote:
Olivier wrote:
2. R Hook-flash key is now available to transfer calls.
2008/10/28 Robert Boardman [EMAIL PROTECTED]
Olivier wrote:
2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hi,
1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP
it is mentioned MWI is now working.
In my testings with lastest 02123
2008/10/29 Brian J. Murrell [EMAIL PROTECTED]
So I have (and have had) jabber configured for some time, specifically
for GTalk, but something has occurred to me. If somebody happens to
send an IM (text) to that account, nobody is going to be receiving it.
I'd like to send a canned message
On Wed, 29 Oct 2008, Peter Evans wrote:
Gordon Henderson wrote:
I just wish there was a fanless version - one feature which I like in the
VIA boards I use.
MSI Wind Board.
No idea about outside Japan, but its fanless, almost certainly
needs convection.
That's because
Olivier wrote:
snip /
I'll reply to the correct thread
[featuremap]
blindxfer = ## ; Blind transfer
;disconnect = *0 ; Disconnect
;automon = *1 ; One Touch Record
atxfer = A ; Attended transfer
so set
On Wed, 29 Oct 2008, Kev Szaszvari wrote:
Hi there
Our company is using the Linksys SPA-942 Phones, and they are pretty useless.
They dont have any central management or provisioning, as well as a pretty
bad interface.
Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have
hi
O use around 500 atcom530, they are work perfect
www.atcom.com.cn
Gordon Henderson wrote:
On Wed, 29 Oct 2008, Kev Szaszvari wrote:
Hi there
Our company is using the Linksys SPA-942 Phones, and they are pretty useless.
They dont have any central management or provisioning, as well as a
Hello!
Just trying to find out how to limit waiting for a RINGING
state for an initiated call by Dial() - This is necessary
since I want to inform the CALLER that destination is not
available if RINGING state was not received within, say 20
seconds. This applies for mostly SIP and IAX2 calls
On Wed, Oct 29, 2008 at 5:46 AM, Peter Evans [EMAIL PROTECTED] wrote:
On Wed, Oct 29, 2008 at 08:45:39AM +, Gordon Henderson wrote:
I wrote:
MSI Wind Board.
No idea about outside Japan, but its fanless, almost certainly
needs convection.
That's because it's called
Gordon,
My guess is that you're a contractor so I can understand why you'd want to keep
yourself in high demand by steering clear of the methods that simplify
deployment and redeployment.
As an employee on the other hand, I want to make things as easy and integrated
as I can in order to
Please help with this strange issue.
When sip show peers returns status Unknown the CDR does not include the
accountcode even though the call is correctly processed.
I'm using A2 Billing and it uses the accountcode to determine the
authentication.
Asterisk version 1.4.21.2
I'm calling from a
Sure it is:
exten = blah,1,Dial(SIP/blah,30)
Where 30 is the time in seconds the application will wait before quitting and
setting the DIALSTATUS variable to NOANSWER.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS -
What options are available for installing an asterisk system onto a
bare-metal system ?
Ones that I have seen:
pbx-in-a-flash
trixbox
astlinux
What I am trying to achieve is to be able to shove a cd / usb into a
machine and have it install asterisk, complete with my .conf files.
I also need
Hi,
I'm trying to access audionativeformat / other codec variables in the hangup
handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response.
Also 'core show channel ...' doesn't list those variables. Are they always set
by asterisk, or only in some scenarios? It's a simple
On Wed, 29 Oct 2008, Steve Totaro wrote:
The power supply probably consumes about as much as the processor!
My question and getting more off topic, but what would one need as far
as battery and solar panels to keep one of these running sans moving
parts?
When I put my Atom board into my
I'm wondering how prevalent the practice of physically segregating voice
and data networks is in the Real World.
What are the factors that typically lead to such a decision?
DIscussions of pros and cons are most welcome by me.
Experiences, anybody?
smime.p7s
Description: S/MIME
On Wed, Oct 29, 2008 at 01:50:05PM +, Julian Lyndon-Smith wrote:
What options are available for installing an asterisk system onto a
bare-metal system ?
Ones that I have seen:
pbx-in-a-flash
Builds from osurce but hides its build scripts. Good luck with fixing
bugs there.
trixbox
On October 29, 2008 10:19:36 am Bill Michaelson wrote:
I'm wondering how prevalent the practice of physically segregating voice
and data networks is in the Real World.
What are the factors that typically lead to such a decision?
DIscussions of pros and cons are most welcome by me.
On Wed, Oct 29, 2008 at 10:25 AM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Wed, Oct 29, 2008 at 01:50:05PM +, Julian Lyndon-Smith wrote:
What options are available for installing an asterisk system onto a
bare-metal system ?
Ones that I have seen:
pbx-in-a-flash
Builds from osurce
On Oct 29, 2008, at 9:19 AM, Bill Michaelson wrote:
I'm wondering how prevalent the practice of physically segregating
voice and data networks is in the Real World.
What are the factors that typically lead to such a decision?
DIscussions of pros and cons are most welcome by me.
Bill Michaelson wrote:
I'm wondering how prevalent the practice of physically segregating
voice and data networks is in the Real World.
What are the factors that typically lead to such a decision?
DIscussions of pros and cons are most welcome by me.
Experiences, anybody?
We chose to go
On Wed, Oct 29, 2008 at 10:32 AM, Andrew Kohlsmith (lists)
[EMAIL PROTECTED] wrote:
On October 29, 2008 10:19:36 am Bill Michaelson wrote:
I'm wondering how prevalent the practice of physically segregating voice
and data networks is in the Real World.
What are the factors that typically lead
On Wed, Oct 29, 2008 at 10:48 AM, Drew Gibson [EMAIL PROTECTED] wrote:
Bill Michaelson wrote:
I'm wondering how prevalent the practice of physically segregating
voice and data networks is in the Real World.
What are the factors that typically lead to such a decision?
DIscussions of pros and
On Wed, 29 Oct 2008, Bill Michaelson wrote:
I'm wondering how prevalent the practice of physically segregating voice and
data networks is in the Real World.
What are the factors that typically lead to such a decision? DIscussions of
pros and cons are most welcome by me.
Customer budget
In my experience most of the serious QoS issues arise in relation to the
Internet pipe (if the provider is IP, and outside the network), not the
LAN. Of course, LANs can be heavily contended, but are not in most
organisations, especially as gigabit cores are getting increasingly
common even
Hi All,
I'm trying to get the user:pass embedded in a SIP Dial string instead
of calling a SIPuser in sip.conf:
Regular way, exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]|30|)
Where the 'sipuser' is a context on sip.conf
[sipuser]
fromuser=sipuser
What I would like to do is embed the
Two separate networks? Did I miss something? I feel like I'm taking crazy
pills! Two separate physical networks means twice the hassle, twice the
maintenance, twice the cost, twice the headache. Not to mention the fact that
the whole idea of VOIP is to simplify IT and focus on converging data
I'm pretty sure they meant two logical networks. At least, I hope they did.
David Gibbons wrote:
Two separate networks? Did I miss something? I feel like I'm taking crazy
pills! Two separate physical networks means twice the hassle, twice the
maintenance, twice the cost, twice the
Did you try show translation
On Wed, Oct 29, 2008 at 3:55 PM, Stanisław Pitucha [EMAIL PROTECTED]wrote:
Hi,
I'm trying to access audionativeformat / other codec variables in the
hangup handler of a call (with ${CHANNEL(audioreadformat)}), but I get no
response. Also 'core show channel
Maybe you have a Codec issue?
On Tue, Oct 28, 2008 at 11:20 PM, Benny Amorsen
[EMAIL PROTECTED][EMAIL PROTECTED]
wrote:
Lincoln King-Cliby [EMAIL PROTECTED] writes:
Periodically I'm seeing calls placed from the 7961s through anything
on the PBX that requires digit entry (the Auto
On Mon, Oct 27, 2008 at 7:12 PM, srinivas Antarvedi
[EMAIL PROTECTED] wrote:
Hello members,
Mysetup:
Asterisk 1.4
Phones:Polycom501
I wanted to register my polycom phones only from a fixed IP(on LAN )
i tried following scenarios and my results are described as follows
1)sip.conf
On Mon, Oct 27, 2008 at 11:53 AM, jordan pan [EMAIL PROTECTED] wrote:
Hi everyone,
I want to use the openser and asterisk to create a system ,who can give
me a detail example about
it,i found it have some complicated.
Thanks in advance.
Does anyone have a recommendation for a headset that plugs into the
Mic/Line-out port on a PC?
Ideally something like the Plantronics SupraPlus. I'd prefer Monaural instead
of stereo, and cheap in price but not in quality.
Thanks for any suggestions...
Jeremy Mann
Director of IT
Texas Health
Benny and Mark,
Thank you for your replies.
I tried adding t1min=500 to sip.conf per the suggestion below and since doing
that haven't been able to reproduce the issue.
If it comes back, I'll do the SIP debug per Mark's suggestion and post the
results here. (Mark, per your question the Auto
I use PostFix and MailHop Outbound from Dyndns.com. They will accept
your outgoing email on multiple ports to help with the blocking
problem. It's $15/year for a limited number of messages.
Todd
On Oct 28, 2008, at 7:39 PM, [EMAIL PROTECTED] wrote:
When I send email from my local
Hello,
I am searched the net for tutorials on how I can Integrate vicidial with
trixbox. I can't find any. Anyone who knows where I can get one?
James
begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
- michel freiha [EMAIL PROTECTED] wrote:
Did you try show translation
That shows a table of times taken by translation... I'm asking about codecs
used by a channel on a certain call.
___
-- Bandwidth and Colocation Provided by
I would contact the vendor.
James Mutuku wrote:
Hello,
I am searched the net for tutorials on how I can Integrate vicidial with
trixbox. I can't find any. Anyone who knows where I can get one?
James
___
-- Bandwidth and Colocation Provided
On Oct 29, 2008, at 8:21 AM, Alex Balashov wrote:
In my experience most of the serious QoS issues arise in relation to
the
Internet pipe (if the provider is IP, and outside the network), not
the
LAN. Of course, LANs can be heavily contended, but are not in most
organisations,
On Wednesday 29 October 2008 10:22:43 David Gibbons wrote:
A phone takes very, very little bandwidth away from the desktop and a decent
one will support tagging its frames for the alternate voice VLAN.
--snip--
In almost all cases it is much better to have two seperate networks.
This may be
I noticed that the vicidial site has documentation available which probably
covers the topics required. However, I also see that they want $50-$100 to
download the docs. Seems harsh.
Ron Byer Jr.
NetWeave Integrated Solutions, Inc.
+1.732.786.8830 x120
-Original Message-
From:
On Tue, 28 Oct 2008, [EMAIL PROTECTED] wrote:
When I send email from my local asterisk machine, my IP address get's
RBL'd.
Asterisk is my only reason for running sendmail, so to keep it simple, I
tried to make my ISP's mail server a 'smart host' (relaying to a trusted
mail server) but my
Daniel Hazelbaker wrote:
I would agree with this as long as you have a decent LAN. We have
about 60 computer workstations and 85 phones on our network. The
entire thing is Gigabit. Each phone (with a few exceptions that we
are running new cable to rectify) has a dedicated ethernet
On Wed, 29 Oct 2008 11:50:31 -0500, Tilghman Lesher wrote
On Wednesday 29 October 2008 10:22:43 David Gibbons wrote:
A phone takes very, very little bandwidth away from the desktop and a decent
one will support tagging its frames for the alternate voice VLAN.
--snip--
In almost all
Hello,
I'm trying to find simple MP3 player in flash, to integrate it with
call recordings.
My requirements would be:
* simple UI
* buffering (would be nice)
* slider
* volume control
* support of 8kHz stereo mp3
* javascript access to seek/position
* free for any use (GPL, MPL, MIT, BSD)
So
David Gibbons wrote:
Two separate networks? Did I miss something? I feel like I'm taking
crazy pills! Two separate physical networks means twice the hassle,
twice the maintenance, twice the cost, twice the headache. Not to
mention the fact that the whole idea of VOIP is to simplify IT and
David Gibbons wrote:
Two separate networks? Did I miss something? I feel like I'm taking crazy
pills! Two separate physical networks means twice the hassle, twice the
maintenance, twice the cost, twice the headache. Not to mention the fact that
the whole idea of VOIP is to simplify IT and
Hi,
For a few weeks now, our asterisk server has been experiencing something
very odd.
From time to time, voicemail.conf would go blank. We finally tracked it
down to happening when someone attempts to change their password.
It seems the file is touched, but not written to, and we're left with
About this mailing:
You are receiving this e-mail because you subscribed to MSN
Kristian Kielhofner wrote:
On Wed, Oct 29, 2008 at 10:48 AM, Drew Gibson [EMAIL PROTECTED] wrote:
Worst part is the few Cisco phones we have insist on searching for
VLAN (which doesn't exist) for 5 minutes on startup. Hopefully they
Drew,
Disable CDP on the phone and that will
On Oct 29, 2008, at 10:10 AM, Darrick Hartman wrote:
David Gibbons wrote:
Two separate networks? Did I miss something? I feel like I'm taking
crazy pills! Two separate physical networks means twice the hassle,
twice the maintenance, twice the cost, twice the headache. Not to
mention the fact
Fair enough, I guess I was concentrating on this line in Jerry's message :)
The only reason I can think of not to is to eliminate the cost of the second
cable.
I believe you're mistaken about the QOS though.
QoS is not required on lightly loaded links and will do nothing for you on
over
After spending a couple hours scanning for an open source (non-
commercial) billing package yesterday I am underwhelmed. Almost all of
the packages listed on the WIKI appear to be defunct, for several
years now. I will be happy to get a login and edit them out if that is
the proper method
I would get a PCAP trace from the phone to see what is going on on the cable.
CS
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Ronald
Wiplinger (Lists)
Gesendet: Dienstag, 28. Oktober 2008 23:01
An: Asterisk Users Mailing List -
Hello,
The paid VICIDIAL user manuals do not cover installing on Trixbox.
Mostly because it can be very difficult to install VICIDIAL on Trixbox
due to the many different versions of Trixbox and the dialplan
complexity of Trixbox.(also I want to mention that there are FREE
versions of the
- Jerry Jones [EMAIL PROTECTED] escribió:
After spending a couple hours scanning for an open source (non-
commercial) billing package yesterday I am underwhelmed. Almost all of
the packages listed on the WIKI appear to be defunct, for several
years now. I will be happy to get a
I do understand that this not free, but BillMax (www.billmax.com)
supports all of your requirements plus includes the source code. I think
you can get a demo that supports under 100 accounts for free... at least
you used to be able to.
j
On Wed, 29 Oct 2008, Jerry Jones wrote:
After spending
Alex Balashov wrote:
Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com
I'm pretty sure they meant two logical networks. At least, I hope they did.
Unfortunately, I was indeed referring to two physical networks. Cabling,
switches, everything, all the way
On Oct 29, 2008, at 12:30 PM, David Gibbons wrote:
Fair enough, I guess I was concentrating on this line in Jerry's
message :)
The only reason I can think of not to is to eliminate the cost of
the second cable.
I believe you're mistaken about the QOS though.
QoS is not required on
Hi,
One of my clients, wants to use * box to run weekly meetings between remote
locations over the internet.
What would be the best configuration for this? We are talking about two
conference rooms.
I am referring to the actual hardware/software and bandwidth requirements
for this to work well.
I
Not directly related to Asterisk but I'm sure one or two of you will get
hot and bothered over this.
:-)
http://deancollinsblog.blogspot.com/2008/10/thecus-n7700.html
Regards,
Dean Collins
Cognation Inc
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 New York
I can think of two valid reasons to physically segregate the networks:
1) Insurance. I.e., to eliminate the possibility that otherwise
properly configured QoS mechanisms become broken, either by
accident, incompetence, or badly-designed or rogue software or
hardware - or are otherwise
On Wed, Oct 29, 2008 at 1:28 PM, Drew Gibson [EMAIL PROTECTED] wrote:
I tried out the cdp-tools some time ago (it may have been on your
recommendation, Kristian) but with no success.
Is it possible to disable CDP on the 7940 (image_version : P0S3-08-2-00)?
regards,
Drew
Hmmm... I guess
Perhaps this is an issue with the SIP registration? Any idea why Asterisk
accepts the call if qualify fails?
Please help with this strange issue.
When sip show peers returns status Unknown the CDR does not include the
accountcode even though the call is correctly processed.
I'm using A2
On Wed, 29 Oct 2008, Robert Augustyn wrote:
Hi,
One of my clients, wants to use * box to run weekly meetings between remote
locations over the internet.
What would be the best configuration for this? We are talking about two
conference rooms.
I am referring to the actual hardware/software
Hi, I have tried installing app_swift on both mac os x and ubuntu now
and am getting the same error. I must be missing something, as I have
tried multiple versions and everytime do sudo make install i get:
if ! [ -f /etc/asterisk/swift.conf ]; then \
install -m 644 swift.conf.sample
Thank you.
What units from Polycom line did you use?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gordon Henderson
Sent: Wednesday, October 29, 2008 4:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Thanks for the reply!
I've played around with R to solve this (probably should have mentioned
that), however I wasn't able to make it work. The message is still played
(this message is from the provider). It will move to the next line in the
dialplan, but as soon as users hear the message they
I've been playing with video phones over the past month or 2.
You've got 3 choices: Bottom-end is Xlite, etc. soft-phones.
Desktop videophones - currently Grandtream GXV3000 and ATL4000's.
Top of the range Polycom video conferencing units.
Starting with the top-of the range
From zapata.conf.sample:
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to work
; with all telcos.
;
; outofband: Signal Busy/Congestion out of band with
I left something out on that last message, sorry.
With r, not R, it will mask the message with ringing. I could then fail it
over to another dial out, however from testing I've found that my users
expect something to happen within 30 seconds (voicemail, pickup, etc.) The
worse-case scenario would
Hi,
I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk
1.4 + CDRTool with freeradius telephony system.
Asterisk is used only for voice mail and redirectioning calls.
Every calls should pass through mediaproxy so that i can account them.
The goal was to create a
Nuno Marques wrote:
Every calls should pass through mediaproxy so that i can account them.
You can do accounting without handling media.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706)
Without mediaproxy? Only based on SIP messages?
2008/10/29 Alex Balashov [EMAIL PROTECTED]
Nuno Marques wrote:
Every calls should pass through mediaproxy so that i can account them.
You can do accounting without handling media.
--
Alex Balashov
Evariste Systems
Web:
Well guys I got it, I started up again making the xml file according to
this:
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP#Downgradingthefirmware
And... voila ! 7911G working with Asterisk and firmware 8.4.0!!! if anybody
need the xml, let me know
Yes. There are some liabilities with that in that the signaling
messages may be incomplete (i.e. you may miss a BYE) and this is the
usual reason given for doing media proxying for more accurate accounting.
But the latency, bandwidth consumption, and increased complexity and
cost associated
Ok... Maybe you're right. I've read somewhere that this service is needed
for taping reasons (policy and other law enforcements). If it's needed whe
can just turn it on for that specific number, right?
But answering to my question, can you point me some ideas refering about
equipment that i
Post it on the wiki! Im sure Ill need it someday
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of César García
Sent: October 29, 2008 6:54 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] XML Cisco config file
Well guys I got it, I started up again making the xml
SIP-only accounting is good enough most of the time.
Does not work in production environment. Specially when you are charging per
second or per minute.
Works only if some one is offering unmetered only service or just doing it
for fun. If it metered service like calling cards, termination or
Jai Rangi wrote:
SIP-only accounting is good enough most of the time.
Does not work in production environment.
Really? Next time I will consult with your authority on what works and
does not work in production environments before implementing for
large-scale billing solutions that are
What version of Asterisk and what version of app_swift?
On 29 Oct 2008, at 15:10, [EMAIL PROTECTED] wrote:
Hi, I have tried installing app_swift on both mac os x and ubuntu now
and am getting the same error. I must be missing something, as I have
tried multiple versions and everytime do sudo
I recently built Asterisk from scratch on Ubuntu (Ubuntu
4.2.3-2ubuntu7). Everything seemed to build ok, but when I start
Asterisk, I get the message:
Warning! Asterisk is not thread safe.
Is this anything to be concerned about? How can I make it go away? Is
there an alternative
Really?
Yes, Specially when your service is metered, I don't know how some once
justify good enough billing. Dealing with 500 customer calling every day for
billing inquiries can turn out to be much more expensive then all other
expenses.
Next time I will consult with your authority on what
By good enough I really did mean good enough, not sort-of kind-of okay.
Jai Rangi wrote:
Really?
Yes, Specially when your service is metered, I don't know how some once
justify good enough billing. Dealing with 500 customer calling every day
for billing inquiries can turn out to be much
Jerry Jones wrote:
After spending a couple hours scanning for an open source (non-
commercial) billing package yesterday I am underwhelmed. Almost all of
the packages listed on the WIKI appear to be defunct, for several
years now. I will be happy to get a login and edit them out if that
On Wednesday 29 October 2008 19:10:49 joe mcguckin wrote:
I recently built Asterisk from scratch on Ubuntu (Ubuntu
4.2.3-2ubuntu7). Everything seemed to build ok, but when I start
Asterisk, I get the message:
Warning! Asterisk is not thread safe.
Is this anything to be concerned about?
Arka:
I thought you would reroute the call with (or without) the leading one, so,
just Dial again.
This will work and your users wont notice a BIG difference if the call is
answered. The problem is if the call is not answer, because if you have a
busy number, then your users will get something
Think more deeply, I understand this is a user forum - but
it doesn not mean that all question must be newbie.
RINGING state meand until I REALLY get a notification from
destination device (SIP for instance) that call have been
accepted by the destination and it have returned
a RINGING -
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