Hello!
You may want to check out http://linuxinnovations.com, a simple
reference describing the practical differences between the various
versions of Asterisk. Seems it includes now version Asterisk 1.8.
--Elliot
On Sun, Nov 14, 2010 at 8:17 AM, Kyle Kienapfel doctor.w...@gmail.com wrote:
Hello!
http://linuxinnovations.com lists the evolution of the various
commands, applications, and other essential parts of Asterisk from
version 1.4 until 1.8, so you may find it a good resource for helping
you make a decision.
--Elliot
--
Hi everyone
I have seen linkedid member in channel structure on 1.8,Actually I
am using 1.6 version, this variable is not there in 1.6 and also I cant
upgrate to 1.8 due to some reason.I wanted to add linked id variable in
channel structure in 1.6 and need to use is CDR as same as in
HI
callee side blind transfer is failed in 1.8 but caller side blind
transfer is succes,Transfer doing by refer method,please help me on this
Nikhil
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Sat, Nov 27, 2010 at 10:51:48AM -0500, Ali Khalfan wrote:
I'm trying to find a way to setup a SIP server that will mainly echo
back a request from one agent only,
Why not use sipp (The package is named sip-tester on Debian/Ubuntu)?
--
Tzafrir Cohen
icq#16849755
HI
I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn)
Out going calls from asterisk to the ata works fine
Incoming calls from the ata to asterisk cuts off with the error msg
Maximum retries exceeded on transmission 70854efe-4157e...@10.168.7.103 for
seqno 102
On Sat, Dec 4, 2010 at 2:11 PM, Shaun Ruffell sruff...@digium.com wrote:
On 12/4/10 9:15 AM, equis software wrote:
HI, I'm using asterisk-1.4.24, dahdi-linux-complete-2.4.0+2.4.0 and
libpri-1.4.11.4
When dial, when 492131 answer, in console appear some error messages
-- AGI
HI
I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn)
Out going calls from asterisk to the ata works fine
Incoming calls from the ata to asterisk cuts off with the error msg
Maximum retries exceeded on transmission 70854efe-4157e...@10.168.7.103 for
seqno 102
Nikhil
Known bug. there is a patch that is in the SVN trunk. I just downloaded the
trunk version last night and will be testing in a bit.
I will keep you posted.
Bryant
From: Nikhil d.nik...@cem-solutions.net
Sent: Monday, December 06, 2010 6:41 AM
To:
Hello list,
I'm trying to set up a video call from my Ekiga client to a Grandstream
GXV3140 IP-phone. The call succeeds but there is no video.
I have in sip.conf :
videosupport=yes
disallow=all
allow=alaw
allow=g726
allow=g729
allow=gsm
allow=h261
allow=h263
allow=h263p
allow=h264
The
On Mon, Dec 06, 2010 at 03:23:42PM +0100, Jonas Kellens wrote:
I'm trying to set up a video call from my Ekiga client to a
Grandstream GXV3140 IP-phone. The call succeeds but there is no
video.
Try restricting video codec to H.261.
R
--
Hello
I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite
on XP as an SIP client:
http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png
The problem is that by default, Asterisk doesn't rewrite the CID name
+ number in incoming calls, so that XLite displays
Le 05/12/2010 20:28, Olivier a écrit :
[...]
Which Dahdi version ?
I had to use latest trunk to have mine working.
Thanks for your reply
SrvPhone2*CLI dahdi show version
DAHDI Version: 2.4.0 Echo Canceller:
FYI I got it: cable was defect.
--
Daniel
--
Here how I changed my information calling an xlite client from a polycom
501.
Sipuser = xlite
144 = polycom
Exten = 145,1,set(CALLERID(num)=5551212)
Exten = 145,n,set(CALLERID(name)=JOES POOL HALL)
Exten = 145,n,Dial(SIP/sipuser,20,m)
-Original Message-
From:
On Sat, 4 Dec 2010, Lee Howard wrote:
I can't pretend to know what an alarm system needs out of a modem, but as far
as iaxmodem acquiring data-modem capabilities that part is already
developing. IAXmodem inherits its DSP capabilities from spandsp, and you can
see on the spandsp that
On Mon, 6 Dec 2010 10:15:34 -0600, Danny Nicholas
da...@debsinc.com wrote:
Here how I changed my information calling an xlite client from a polycom
501.
Sipuser = xlite
144 = polycom
Exten = 145,1,set(CALLERID(num)=5551212)
Exten = 145,n,set(CALLERID(name)=JOES POOL HALL)
Exten =
On Mon, 06 Dec 2010 20:03:03 +0100, Gilles codecompl...@free.fr
wrote:
Any idea why Asterisk shows nothing, and how to retrieve the original
CID information?
Sorry about that, I forgot that the console had to be started in
verbose mode for NoOp() to display data:
asterisk -r
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Monday, December 06, 2010 1:23 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [3102] How to rewrite CID name + number?
On Mon,
On 12/06/2010 11:21 AM, Jeff LaCoursiere wrote:
Ah, but it is not for me personally, it is for the ~20,000 residents of the US
Virgin Islands that currently pay $40/month for a POTS line just for their
alarm monitoring. As far as the reliability of POTS lines - in our area, that
is exactly
What you probably have is a DSL MODEM that can act as a ROUTER but most
likely doesn't have to.
Your device probably has the same capabilities as most modems, the added
features of NAT, DHCP, and whatever else. Normally you can disable that
additional functionality. Now you just have a DSL
Dear Steve;
Really until now, I am not able to know if Vyatta has a DSL router (hardware)
that can be used to do the QoS and bandwidth management without need to
download the software of Vyatte and install at the server?
I am trying actually not to let all the traffic passing Asterisk server
If I were you I would visit their site! I seriously doubt that they have a
DSL router. They came out with appliances, maybe they do. Go empower
yourself and look at their offerings.
The first thing they put out as an appliance was a Dell R200. That was cool
because we used Dell R200s in our
On Mon, 6 Dec 2010 13:39:33 -0600, Danny Nicholas
da...@debsinc.com wrote:
#2 you might want to save the original ID to a variable, the reset
CALLERID(num) to that variable. (if #2 is corrected, this one probably won't
matter).
Thanks Danny, and sorry for the trouble: I was paying so much
According to the Admin guide EFK is not supported on 501s
This capability applies to the SoundPoint IP 32x/33x, 450, 550, 560, 650,
and
670 desktop phones, the SoundStation IP 5000, 6000, and 7000 conference
phones, and Polycom VVX 1500 business media phones
On Fri, Dec 3, 2010 at 5:02 PM,
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gord Urquhart
Sent: Monday, December 06, 2010 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Park by EFK
According
On 12/05/2010 08:25 AM, C F wrote:
Is there any version matching doc? since it was changed to Dahdi I
don't really know which version works with which.
Asterisk 1.4.21 and lower can only use Zaptel. Asterisk 1.4.22 through
1.4.x can use either Zaptel or DAHDI. Asterisk 1.6.x, Asterisk 1.8.x
Hi, i have context in a dialplan, I want to execute this context without
insert the Answer Application (sò ..without call any ext).
Example :
[sistema-allarmi-principale]
exten = s,1,Set(GRUPPO=${DIAL:-2:1})
exten = s,2,Set(ALLARME=${DIAL:1:1})
exten = s,3,AGI(checkgroup.php|${GRUPPO})
;rest
Thanks Kevin.
Upto which version fo Dahdi works with 1.4.x?
On Mon, Dec 6, 2010 at 6:25 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 12/05/2010 08:25 AM, C F wrote:
Is there any version matching doc? since it was changed to Dahdi I
don't really know which version works with which.
I am trying to dial through my asterisk machine from phone A to phone B.
My DID is registered properly with the SIP provider. When I dial from
A to B it looks fine so far.
A rings B and B can pick up and the call is bridged. However, I don't
hear any audio so therefor it is not working.
I am
Yes sir,
We are pass the error. Works like a charm. I just documented this on our
new wiki:
http://voipcomsolutions.com/wiki/index.php?title=How_to_install_Asterisk_from_source_-_Google_Integration_ready
Thanks again
*José Pablo Méndez
*
2010/12/1 José Pablo Méndez Soto
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