Tzafrir Cohen wrote:
On Sun, Jan 20, 2008 at 06:33:31PM +1100, Rob Hillis wrote:
PC's age and when they age, things tend to go wrong, particularly when
you upgrade software. Unusual crashes are usually the first sign that
something is going wrong.
And suddenly the same PC has unaged when
Tilghman Lesher schreef:
On Saturday 19 January 2008 20:32:42 Michael J. Liberatore wrote:
Hi i have a friend who i setup an asterisk system for at his doctors
office. it has 3 snom 360 phones with 6.2.x stable firmware and latest
asterisk 1.4 and zaptel. They have the digium 4 port fxo
Hi All;
Anyone can advise for a good IP Phone that has the
ability to support SIP firmware and IAX firmware?
Ofcourse, SIP there is a lot, but we need also the
ability to use IAX (as it is good for NAT).
Any advise.
Regards
Bilal
Not the first time I've seen something like this happen. If you read
what I said, I wasn't saying that this /was/ what was happening with his
hardware, merely that it's the first sign.
Tzafrir Cohen wrote:
On Sun, Jan 20, 2008 at 06:33:31PM +1100, Rob Hillis wrote:
PC's age and when they
Okay, so this time I wont send the email to the bounce address :-)
Can I include an alsa channel in a Page() channel list?
Is there a way I can have an extension call a group of phones and put
them into a pre-existing conference (muted) while the caller goes into
the conference (unmuted, and
On Jan 19, 2008 9:26 PM, Rob Hillis [EMAIL PROTECTED] wrote:
I wasn't intending to blame Ira for his own problems - I was intending to
point out that running a production system on discarded hardware is a really
bad idea.
I wasn't even suggesting a mammoth server - as you may or may not
Hi All;
Anyone can advise for a good IP Phone that has the
ability to support SIP firmware and IAX firmware?
Ofcourse, SIP there is a lot, but we need also the
ability to use IAX (as it is good for NAT).
Any advise.
Regards
Bilal
I am using an atcom at-530
Tzafrir Cohen wrote:
Well, there is not enough data to suggest that. Before blaming Ira for
being such a cheap fellow (after all, he didn't buy one of those IBM big
iorns to run Asterisk on) we should also consider that the upgrade to
1.4 probably also involved an upgrade of Zaptel, which
Thomas Kenyon wrote:
Tzafrir Cohen wrote:
Well, there is not enough data to suggest that. Before blaming Ira for
being such a cheap fellow (after all, he didn't buy one of those IBM big
iorns to run Asterisk on) we should also consider that the upgrade to
1.4 probably also involved an
Hi All;
I tried Firefly softphone with IAX and it gave very
poor quality.
Any one advise a good strong softphone that can work
with IAX fine?
Regards
Bilal
Be a better friend, newshound, and
Hi All;
Did anyone try to use IAX IP Phone behind NAT, and let
it receive calls from Asterisk without doing port
mapping at the router existed at the site where the
IAX IP Phone existed? Is the need just to let the IAX
IP Phone that is NATed to register on the Asterisk and
at asterisk I set
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fons van
der Beek
Sent: Sunday, January 20, 2008 3:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls Being Randomly Bridged
Tilghman Lesher schreef:
On
Try zoiper
bilal ghayyad ha scritto:
Hi All;
I tried Firefly softphone with IAX and it gave very
poor quality.
Any one advise a good strong softphone that can work
with IAX fine?
Regards
Bilal
On Sun, 2008-01-20 at 13:26 +1100, Rob Hillis wrote:
I wasn't intending to blame Ira for his own problems - I was intending
to point out that running a production system on discarded hardware is
a really bad idea.
Let me jump in on that.
Some other posters mention (un-)aging of systems.
All
On 20:43, Sat 19 Jan 08, Russell Bryant wrote:
Matthew Rubenstein wrote:
I'd be even more likely to use nightly (or other periodic snapshot,
even weekly) .deb packages. Because then I could use APT to notify me
and manage them. Especially if they included a changelog (which APT
Michael J. Liberatore schreef:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fons van
der Beek
Sent: Sunday, January 20, 2008 3:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls Being
On 13:06, Sun 20 Jan 08, Fons van der Beek wrote:
Michael J. Liberatore schreef:
On the snom 360
If you pay close attention when you transfer the calls, you can see the
names/numbers of the calling partners
by using the cursor button (the round button with arrows) you can
select to who you
cancallforward: yes
setvar:
Any help would be appreciated.
Regards,
Atis
The relevant portion of UPGRADE.txt mentions that a call-limit is
necessary in
order for SIP devices to report proper device state. I see in your
sip.conf file
that you have set call-limit in the general
On Sun, 20 Jan 2008, bilal ghayyad wrote:
Hi All;
Did anyone try to use IAX IP Phone behind NAT, and let
it receive calls from Asterisk without doing port
mapping at the router existed at the site where the
IAX IP Phone existed? Is the need just to let the IAX
IP Phone that is NATed to
i want to use Cisco AS5400 media Gateway as my PSTN Gateway instead of using
the E1 PCI cards in asterisk box ,is this practically possible? can i use SIP
in the connection between Asterisk and Cisco AS 5400 Gateway?
_
Express
we are experiencing 30 second delay before voice is heard after answer
when we ran wireshark it showed the problem
between frames 634 (where the softphone answers)
and 1366. Between those frames, asterisk receives RTP packets from
both the softphone and the sip carrier, but doesn't forward them
I have been running 1.4.17 since its release, and no kernal panics.
Before that I was running 1.4.13 without any kernal panics.
System Specs:
4 Core Xeon 5110 @ 1.6Ghz (two dual proc chips)
8 Gb Ram
400GB Raid 5 SAS Array
-- Original Message --
Any one advise a good strong softphone that can work with IAX fine?
samelessplugTry my softphone :
http://www.marccharbonneau.com/asterisk/mediaxphone.php/samelessplug
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On Jan 20, 2008 11:10 AM, love U. all [EMAIL PROTECTED] wrote:
i want to use Cisco AS5400 media Gateway as my PSTN Gateway instead of
using the E1 PCI cards in asterisk box ,is this practically possible?
Yes
can i use SIP in the connection between Asterisk and Cisco AS 5400 Gateway?
Of
At 11:33 PM 1/19/2008, you wrote:
PC's age and when they age, things tend to go wrong, particularly
when you upgrade software. Unusual crashes are usually the first
sign that something is going wrong.
Well, my experience is they work until they die and that's usually
the PS or HD. In that
Hello there,
I just wasted some time setting up a Grandstream HT-488 to be used with
Asterisk, so I thought I'd share the experience by writing a small
tutorial at: http://astrecipes.net/index.php?n=338
Most tutorials I came across were for old versions of the firmware, and I
spent too
On 15:10, Sun 20 Jan 08, Ira wrote:
At 02:59 PM 1/20/2008, you wrote:
I was getting kernel panics from HPEC, but it was because I was using
the i386 binary and not the i686 one.
I called Digium, they logged in, sorted it out, and everything works
fine now.
I wonder if that's my problem?
At 02:59 PM 1/20/2008, you wrote:
I was getting kernel panics from HPEC, but it was because I was using
the i386 binary and not the i686 one.
I called Digium, they logged in, sorted it out, and everything works
fine now.
I wonder if that's my problem? I have a 1ghz Celeron and I think I'm
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ira wrote:
At 02:59 PM 1/20/2008, you wrote:
I was getting kernel panics from HPEC, but it was because I was using
the i386 binary and not the i686 one.
I called Digium, they logged in, sorted it out, and everything works
fine now.
I wonder
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Hash: SHA1
William Stillwell (Ki4swy) wrote:
I have been running 1.4.17 since its release, and no kernal panics.
Before that I was running 1.4.13 without any kernal panics.
I was getting kernel panics from HPEC, but it was because I was using
the i386
I have added two extentsions. I am try to test connecting X-lite to the
server.
I have two extension one 1000 with password 1234 and one 2000 with
password 2000.
I have the softphone on the same network so I do not have to worry about
ports being open.
So I have in the properties of Account
You could also use a call file ( google it ;) ).
On Jan 18, 2008 6:37 PM, LWATCDR [EMAIL PROTECTED] wrote:
I would like to add a function to an existing application that will
make an outgoing call.
I found this example using the Manager API for originating a call to
an extension.
Does Asterisk give you any feedback on the console?
On Jan 21, 2008 12:14 PM, Andrew Ladanowski
[EMAIL PROTECTED] wrote:
I have added two extentsions. I am try to test connecting X-lite to the
server.
I have two extension one 1000 with password 1234 and one 2000 with password
2000.
I
On Jan 20, 2008 7:14 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote:
I have added two extentsions. I am try to test connecting X-lite to the
server.
I have two extension one 1000 with password 1234 and one 2000 with password
2000.
Andrew - could you send us the relevent sections of your
No
Andrew Ladanowski
AddInSolutions Inc.
www.addinsol.com
[EMAIL PROTECTED]
Phone: 954-815-2402
Fax: 954-414-8432
CONFIDENTIAL : The information in this email (including any attachments) is
confidential and may be privileged. If you are not the intended recipient, you
may not and must not
Here are my log information.
[Jan 20 12:34:00] NOTICE[2637] chan_sip.c: Registration from
'Andrewsip:[EMAIL PROTECTED]' failed for '192.168.3.116' - Device does not
match ACL
[Jan 20 12:35:33] NOTICE[2637] chan_sip.c: Registration from
'Andrewsip:[EMAIL PROTECTED]' failed for '192.168.3.116'
1. SSH Into the server
2. cd /etc/asterisk/
3. cat sip.conf
and copy and paste the output here
Regards
Kev
Andrew Ladanowski wrote:
Here are my log information.
[Jan 20 12:34:00] NOTICE[2637] chan_sip.c: Registration from
'Andrewsip:[EMAIL PROTECTED]' failed for '192.168.3.116' - Device
Hello Andrew,
On Jan 21, 2008 9:14 AM, Andrew Ladanowski [EMAIL PROTECTED]
wrote:
I have added two extentsions. I am try to test connecting X-lite to the
server.
- - - s n i p - - -
What desktop OS are you using on your X-Lite? If you are using Mac OS X
Leopard and even the latest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel
van Baak
Sent: Sunday, January 20, 2008 7:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Calls Being Randomly Bridged
On 13:06, Sun 20 Jan 08, Fons van der Beek wrote:
I actually have both running in two different virtual pc windows and I have the
same problem.
Andrew Ladanowski
AddInSolutions Inc.
www.addinsol.com
[EMAIL PROTECTED]
Phone: 954-815-2402
Fax: 954-414-8432
CONFIDENTIAL : The information in this email (including any attachments) is
Windows XP.
Thanks
Andrew Ladanowski
AddInSolutions Inc.
www.addinsol.com
https://mymail.ladanowski.com/exchweb/bin/redir.asp?URL=http://www.addi
nsol.com
[EMAIL PROTECTED]
Phone: 954-815-2402
Fax: 954-414-8432
CONFIDENTIAL : The information in this email (including any
Hi All,
I'm not sure if this is related directly to asterisk or not but on my
Polycom 320 when I try to dial a number smaller than 4 digits I get an
error on the phone saying Enter more digits.
The dial plan section is listed below.
dialplan dialplan.impossibleMatchHandling=0
On Jan 20, 2008 7:47 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote:
Here are my log information.
[Jan 20 12:34:00] NOTICE[2637] chan_sip.c: Registration from
'Andrewsip:[EMAIL PROTECTED]' failed for '192.168.3.116' - Device does
not match ACL
[Jan 20 12:35:33] NOTICE[2637] chan_sip.c:
steve,
thanks for posting this tidbit!
daveC
Steve Johnson wrote:
Sorry to answer my own post, but I have found a solution which perhaps
others can use too...
In the .call file, instead of specifying a channel line as:
chan: SIP/140 (for example)
use instead:
chan:
On Jan 20, 2008 8:06 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote:
Windows XP.
Andrew - you're going to need to get us your sip.conf before we can
really assist you any further.
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Actually I got it to work with admin not root.
Thanks for trying.
Andrew Ladanowski
AddInSolutions Inc.
www.addinsol.com
[EMAIL PROTECTED]
Phone: 954-815-2402
Fax: 954-414-8432
CONFIDENTIAL : The information in this email (including any attachments) is
confidential and may be privileged. If
Which Linux distribution are you using?
SSH for root might be denied in your setup
On Jan 21, 2008 1:29 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote:
I can not exit out of my Asterisk set up it. When I try to login to my
server using ssh in denies the username and password. I assume the
root login is not permitted by default via ssh
Try the username admin and the password you set during the install
Andrew Ladanowski wrote:
I can not exit out of my Asterisk set up it. When I try to login to my
server using ssh in denies the username and password. I assume the default
name
I can not exit out of my Asterisk set up it. When I try to login to my server
using ssh in denies the username and password. I assume the default name was
root when I set up the Asterisk. I remember the password.
Andrew Ladanowski
AddInSolutions Inc.
www.addinsol.com
[EMAIL PROTECTED]
Phone:
Basically, You will need to send the sip.conf file. It will not work
unless you have stuff set up in sip.conf.
x-Lite works fine; I'm using it without a hitch.
HTH,
Shane
On 1/20/08, Erik Anderson [EMAIL PROTECTED] wrote:
On Jan 20, 2008 8:06 PM, Andrew Ladanowski [EMAIL PROTECTED]
wrote:
Sorry for the delay I needed to ftp to my windows server
; SIP Configuration example for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/[EMAIL PROTECTED] to call any SIP user on
Fons van der Beek schreef:
Hello
Is het possible to assign blf to a misdn channel?
I want to watch the status of my external misdn channels on a linksys
962, e.g. green = available , red = in use
and as an extra I want when I press the blf use the external line or
when busy i want to barge
Next Question What pin holes are required to run this?
Andrew Ladanowski
AddInSolutions Inc.
www.addinsol.com
[EMAIL PROTECTED]
Phone: 954-815-2402
Fax: 954-414-8432
CONFIDENTIAL : The information in this email (including any attachments) is
confidential and may be privileged. If you are not
The granstream gxp-2000 has the blf/line buttons but they are terrible
phones.
Am I missing any phones? Any other suggestions?
I have to agree with your point - the transfer on the Snom's is not good
if you have to juggle several calls. The Polycom transfer system is
probably the best,
Sorry everyone. There was an error in the dial plan in Asterisk.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Klaverstyn, David C
Sent: Monday, 21 January 2008 12:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom 320 Issue
Hello
Is het possible to assign blf to a misdn channel?
I want to watch the status of my external misdn channels on a linksys
962, e.g. green = available , red = in use
and as an extra I want when I press the blf use the external line or
when busy i want to barge in that call.
Did somebody do
Hello,
I am having a lot of trouble with my deployment of Asterisk. I am running
the PBX-In-a-flash turnkey of Asterisk and ever since deployment I have had
many different problems. I have managed to get all issues sorted out as I go
along, until this one that randomly began last week.
We are
Yes... and there is plenty of information about sip-to-sip communications
if you do research
CS
i want to use Cisco AS5400 media Gateway as my PSTN Gateway instead of
using the E1 PCI cards in asterisk box ,is this practically
possible? can i use SIP in the connection between Asterisk and
www.zoiper.com
We have been using this one, not like X-Lite but it works well.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Generally, E1 is pretty rock solid so my guess is more inside the
network.
We found an issue at a site a while ago which was pretty bad (calls
cutting off randomly) and we fixed it by disconnecting the voice and
data networks. We could have troubleshot it properly, but fitting an
extra network
You can try Zoiper.follow the given link
http://www.zoiper.com/
--Keshav
bilal ghayyad [EMAIL PROTECTED] wrote: Hi All;
I tried Firefly softphone with IAX and it gave very
poor quality.
Any one advise a good strong softphone that can work
with IAX fine?
Regards
Bilal
On Jan 20, 2008 11:04 PM, Cameron Hissey [EMAIL PROTECTED] wrote:
Hello,
I am having a lot of trouble with my deployment of Asterisk. I am running
the PBX-In-a-flash turnkey of Asterisk and ever since deployment I have had
many different problems. I have managed to get all issues sorted out
On Mon, Jan 21, 2008 at 12:23:06AM +0100, Michiel van Baak wrote:
On 15:10, Sun 20 Jan 08, Ira wrote:
At 02:59 PM 1/20/2008, you wrote:
I was getting kernel panics from HPEC, but it was because I was using
the i386 binary and not the i686 one.
I called Digium, they logged in, sorted it
There is a small one, called iaxLite, developed by my friend.
http://iaxtalk.com/index.php?main_page=product_infocPath=6products_id=7
gt
2008/1/21, Keshav K. [EMAIL PROTECTED]:
You can try Zoiper.follow the given link
http://www.zoiper.com/
--Keshav
bilal ghayyad [EMAIL PROTECTED]
Hi,
I'm trying to install asterisk-addons-1.6.0-beta1 on my machine.
But getting following error during make:
[EMAIL PROTECTED] asterisk-addons-1.6.0-beta1]# make
make[1]: Entering directory `/usr/src/asterisk/asterisk-addons-1.6.0-beta1'
[CC] app_addon_sql_mysql.c - app_addon_sql_mysql.o
2008/1/21, Michael J. Liberatore [EMAIL PROTECTED]:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel
van Baak
Sent: Sunday, January 20, 2008 7:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Calls Being Randomly
At 03:23 PM 1/20/2008, you wrote:
To make sure what you are running issue an 'uname -a'
The Celeron is i686.
It says some stuff followed by: i686 i686 i386
Ira
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