On Wed, 2012-11-14 at 09:48 -0800, Michael L. Young wrote:
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 14, 2012 9:25:37 AM
Subject: Re:
On Thu, 2012-11-15 at 08:52 +0100, Olivier wrote:
When a Polcyom SoundPoint gets a VLAN ID from a DHCP server, does it
store this VLAN ID into its flash memory so that, on next boot, it
would broadcast its DHCP request using the VLAN he previously got ?
As far as I know it doesn't. It
On Thu, 2012-11-15 at 08:53 +0100, Hans Witvliet wrote:
In stead of 12345678 i would like to use b.c.o.gr...@minoss.nl
But afaicr the dots will cause problems
If your extension does not start with an underscore, it is not
considered as an extension pattern. Correct me if I'm wrong please!
2012/11/15 Frederic Van Espen frederic...@gmail.com
On Thu, 2012-11-15 at 08:52 +0100, Olivier wrote:
When a Polcyom SoundPoint gets a VLAN ID from a DHCP server, does it
store this VLAN ID into its flash memory so that, on next boot, it
would broadcast its DHCP request using the VLAN he
Hello,
I am using asterisk 1.6.x and 1.8.18.0 (LTS) on a CentOS 5 boxes. I'm using
php AGI for incoming calls and after an hour of running, there is a
mismatch b/w active channels (ss7) and active calls in core show channels
count. Active calls are much higher than even the physical SS7 channel
2012/11/15 Frederic Van Espen frederic...@gmail.com
On Thu, 2012-11-15 at 08:53 +0100, Hans Witvliet wrote:
In stead of 12345678 i would like to use b.c.o.gr...@minoss.nl
But afaicr the dots will cause problems
If your extension does not start with an underscore, it is not
considered
Hello,
Does anyone know if it's possible to setup the following scenario?
1. A specific ext(let's say 111) is on active call with an external number
via SIP (let's say 22334455).
2. Via a web GUI, send to asterisk another phone number (22556677) and the
ext number (111).
3. Asterisk initiates a
On 15 Nov 2012, at 14:21, Michael wrote:
Hello,
Does anyone know if it's possible to setup the following scenario?
1. A specific ext(let's say 111) is on active call with an external number
via SIP (let's say 22334455).
2. Via a web GUI, send to asterisk another phone number (22556677)
On Thu, Nov 15, 2012 at 10:11:20AM +0530, Harish Mandowara wrote:
Dear,
i using this scenario.
jitsi--- asteriskEPABX-- Local Telephone
What DAHDI device is used for the connection 'asterisk - EPABX'? A
digital one? (PRI?) or an analog one? (FXO?)
when i am calling from jitsi to
At present I have two hardware identically freepbx/asterisk boxes. The
mysql db on one is slaved to the other and all config files are
rsync'd once every 24 hours (we have few configuration changes).
We use Polycom 321/331/550/650 phones, and I notice that these phones
can be configured with two
On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger
cnighswon...@foundations.edu wrote:
At present I have two hardware identically freepbx/asterisk boxes. The
mysql db on one is slaved to the other and all config files are
rsync'd once every 24 hours (we have few configuration changes).
We
Polycom phones after firmware 2.x register to BOTH the primary and backup
servers.
On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger
cnighswon...@foundations.edu wrote:
Would the simplest approach to failover be to just configure my
primary asterisk server as the
You can actually configure at least some Polycom phones to 3 or more SIP
servers. Your problem is going to be that when one of your servers is down
for whatever reason, the line key attached to that server will be off.
In a Dual Server environment, I would lean toward putting something like
On 11/15/2012 10:27 AM, Eric Wieling wrote:
What I have found most difficult in any failover situation is having
everything decide at the same time something has failed.
(this applies to anything not just asterisk)
For example how does the polycom react if it can make the sip
connection, but
On 11/15/2012 10:31 AM, Danny Nicholas wrote:
ran into this before on routers, you can put something like that or vrrp
or carp in front of a pair of systems to fail to the right one BUT there
isn't only one interface on something like a pbx, it has a lan interface
and a wan interface, you
Hey, all. I'm interested in doing some simple, very specific web pages
for some of my users -- things like call groups, setting forwarding, and
for the receptionist to transfer calls and see calls. Probably do this
in Ruby or PHP, though I'm open-minded. Anyway, if someone could point
me to
On Nov 15, 2012, at 10:54 AM, Ken D'Ambrosio wrote:
Hey, all. I'm interested in doing some simple, very specific web pages for
some of my users -- things like call groups, setting forwarding, and for the
receptionist to transfer calls and see calls. Probably do this in Ruby or
PHP,
On Wed, Nov 14, 2012 at 7:42 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Nov 2012, Face wrote:
Is there a way I can trigger a AGI script On SIP REGISTER event.
On Wed, 14 Nov 2012, Danny Nicholas wrote:
What you will need to do is to monitor for the SIP REGISTER in AMI,
Heh. Shortly after I sent my e-mail, I bumped into the Adhearsion you
mentioned, below. Boy, but that looks exactly like what I'm thinking
of! Thanks much...
-Ken
On 2012-11-15 13:08, David M. Lee wrote:
On Nov 15, 2012, at 10:54 AM, Ken D'Ambrosio wrote:
Hey, all. I'm interested in
Those are asterisk downloads, not dahdi downloads
Justin Killen
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Oyarzo
Sent: Wednesday, November 14, 2012 4:28 PM
To: Asterisk Users Mailing List -
SIP peer doesn't need to be registered. It uses the host value to talk to
the peer.
Regards,
Ali Pey
On Thu, Nov 15, 2012 at 1:15 PM, Face falaz...@gmail.com wrote:
On Wed, Nov 14, 2012 at 7:42 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Nov 2012, Face wrote:
Is there
On 12-11-13 11:38 PM, bilal ghayyad wrote:
Dears;
What Jian said is the right and it worked.
But I have the following questions:
Why 192.168.10.2 is wrong and I have to use 192.168.10.0? Also, do I have to
set the localnet or it is enough to set the externip?
The IP information is called
On Monday, November 19th, 2012, the Asterisk community services
listed below will be undergoing maintenance (software upgrades and
updates). The services will be shut down at approximately 9:00 PM CST
(3:00 AM November 20th UTC), and will return no later than 10:00 PM
CST. We apologize in advance
I am running 1.4.43
Trying to use AGI to do the
Application: AGI
Its telling me missing action in request.
What should the Action: be in this case
I tried
Action: Originate
but it still says the same error.
--
_
--
While AGI is an application, it has to be done within a call, so it is best
to do a call to the context that has the AGI command in it.
[runmyagi]
Exten = s,1,AGI...
Command: Dial
Context: runmyagi
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Check for Status on these commands. If it comes back OK the peer is
registered. If not, it should return UNKNOWN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Face
Sent: Thursday, November 15, 2012 12:15
hile AGI is an application, it has to be done within a call, so it is best
to do a call to the context that has the AGI command in it.
[runmyagi]
Exten = s,1,AGI...
Command: Dial
Context: runmyagi
Danny
Thanks, what might the command look like to add one asterisk box (sip
connected)
to an
Exten = s,1,meetme(6500,I,1234)
Or
Exten = s,1,MeetMe(6500,Mrsqx)
Is what I use to connect to conference 6500. This starts the conference if
you are caller 1 or adds if you are caller N.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On Thu, 2012-11-15 at 12:13 +0100, Frederic Van Espen wrote:
On Thu, 2012-11-15 at 08:53 +0100, Hans Witvliet wrote:
In stead of 12345678 i would like to use b.c.o.gr...@minoss.nl
But afaicr the dots will cause problems
If your extension does not start with an underscore, it is not
Up one level are the DAHDI directories
http://downloads.asterisk.org/pub/telephony/
JN
Justin Killen wrote:
Those are asterisk downloads, not dahdi downloads
Justin Killen
I think you can use virtual IP with a group of sip server that run in HA .
so, only one of sipserver is handle the call and other is standby ...
Is this what you want ...?
Regard/chui king man
寄件人︰ Danny Nicholas da...@debsinc.com
收件人︰ 'Asterisk Users Mailing List - Non-Commercial Discussion'
It is needed to purchase . Any other option that is open source and free ???
寄件人︰ Michelle Dupuis mdup...@ocg.ca
收件人︰ kingman chui chuiking...@yahoo.com.hk; Asterisk Users List
asterisk-users@lists.digium.com
傳送日期︰ 2012年11月16日 (週五) 8:08 AM
主題︰ RE: [asterisk-users] 回覆︰ Simple failover
On Fri, Nov 16, 2012 at 12:32 AM, Danny Nicholas da...@debsinc.com wrote:
Check for Status on these commands. If it comes back OK the peer is
registered. If not, it should return UNKNOWN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I try before , I connect AMI by telnet localhost ..
In this session , if I register a softphone or unregiser from softphone ,
There is event message and show peer no : XXX ,and show register or unregister .
You can capture this message and do further what you want ...
Regad/chui king man
Hello,
After Upgrade to Asterisk 11.1.0-rc1 I keep getting
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [603@DLPN_AlDimnaDialPlan:601]
Dial(SIP/601-0002, SIP/603) in new stack
[Nov 16
also sprach Shaun Ruffell sruff...@digium.com [2012.11.08.1615 +0100]:
My systems are already managed automatically, thankfully no longer
with Puppet. ;)
Just out of curiosity why do you say this?
Sorry for the late reply, I don't want to go into this on the list,
but if you are curious:
also sprach Paul Belanger paul.belan...@polybeacon.com [2012.11.08.2304
+0100]:
Either way, it sounds like you need to store your data some place and
start building it out.
To recap: given that Asterisk RealTime doesn't really provide
anything more than real-time access to data (i.e. the data
Hi you can get some help using n-way dialplan example. Its generate new
call and transfer current call in conference meetme. You can google to find
its example
On Nov 15, 2012 8:15 PM, Michael voip.quest...@gmail.com wrote:
Hi Aldo,
Thank you very much for answering my question.
Can you
Hi,
Check your php AGI properly , also check it without dataabse and AGI, may
be your database creating issue.
On Thu, Nov 15, 2012 at 5:51 PM, [Digital^Dude] ®
millennium@gmail.comwrote:
Hello,
I am using asterisk 1.6.x and 1.8.18.0 (LTS) on a CentOS 5 boxes. I'm
using php AGI for
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