--- On Thu, 2/10/11, Jonathan Thurman wrote:
> Have you looked at the 'defaultip' sip configuration
> option? Or
> setting host= for those devices?
I've read that defaultip can only be used on type=peer and when host=dynamic.
I use type=friend.
host=IP seems to be OK for me.
I actually trie
On Wed, Feb 9, 2011 at 6:55 AM, Vieri wrote:
[snip]
> Since all of the SIP devices in my LAN have static IP addresses, I can keep
> track of
> everyone on my own. For instance, could I do "fake" SIP registrations from
> localhost
> (the * server) and specify a LAN IP address?
Have you looked
On 2/9/11 6:55 AM, Vieri wrote:
I'd like to do that without Realtime (or with Realtime+FreePBX) or with any
other means that doesn't require more than 2 servers (2 asterisk boxes)?
we use drbd & nfs cluster to store asterisk's ASTDB & voice mail
files but that would involve installing 2 extra
--- On Tue, 2/8/11, Jonathan Thurman wrote:
> It depends on your configuration. If you use Asterisk
> Realtime to
> store SIP registrations, then the database will contain
> information on
> how to contact the device (fullcontact, ipaddr, and port
> fields).
> Then on a failover, Asterisk will
On Tue, Feb 8, 2011 at 8:07 AM, Vieri wrote:
> Suppose you have 2 identical Asterisk servers and 1 alias IP address that you
> assign to either one, according to system failures, etc.
> Also suppose that all SIP clients register requests go to the alias IP
> address.
This is a typical setup for
Hi,
Thats very simple.
Use sip realtime registration with mysql and heartbit to control switiching.
Regards,
Carlos M Cruz
Em 2011/02/08 16:07, "Vieri" escreveu:
Hi,
Suppose you have 2 identical Asterisk servers and 1 alias IP address that
you assign to either one, according to system failu
users-boun...@lists.digium.com] On Behalf Of Gergo Csibra
[csi...@gmail.com]
Sent: Tuesday, February 08, 2011 11:17 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] fail-over server
Tuesday, February 8, 2011, 5:07:29 PM, Vieri wrote:
> How can I minimize this time lapse? Can Asterisk "notify&quo
Tuesday, February 8, 2011, 5:07:29 PM, Vieri wrote:
> How can I minimize this time lapse? Can Asterisk "notify" all SIP
> clients in its sip.conf that they need to acknowledge being on-line
> or not (thus forcing re-registration in my scenario)?
If you have two identical servers online, it is bet
Hi,
Suppose you have 2 identical Asterisk servers and 1 alias IP address that you
assign to either one, according to system failures, etc.
Also suppose that all SIP clients register requests go to the alias IP address.
Imagine server1 fails and server2 gets the alias IP address. Correct me if I'
All,
Thanks for the help. Checking on and changing the route based on
dialstatus is the way to go.
Thanks,
___
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> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Sent: Friday, January 27, 2006 2:45 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connection
fail
On Friday 27 January 2006 16:24, Damon Estep wrote:
> If you have qualify=yes I assume that triggers a sip query to get
> channel capabilities from the peer? What is the qualify timeout? Can it
> be manipulated?
qualify (for SIP) sends a SIP OPTIONS packet to the peer and waits for a
response. I
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Sent: Friday, January 27, 2006 2:07 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connecti
On Friday 27 January 2006 16:00, Damon Estep wrote:
> In the event that the first attempt DOES NOT RESPOND (is down) there has
> to be a timeout value to go to the next priority, correct? Otherwise the
> channels just sits silent waiting for a response.
That's what the qualify parameter in sip/iax
xecuted?
Damon
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Sent: Friday, January 27, 2006 1:12 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP conne
On Thursday 26 January 2006 10:52, Cavanna, Richard wrote:
> I am trying to tweak my dial plan and I am running into a problem.
> Sometimes my VoIP out bound calls do not complete on overseas calls(busy
> or just a hang-up). Is there a way in the dial plan to automatically
> dial out of my PRI whe
I know this may be a backwards way but for several
reasons I have asterisk send all calls thru astcc.
With astcc you specify multiple routes with prioroty
settings. If it cant complete a call with one route it
will roll over and use the next one.
Regards,
Dovid
--- "Cavanna, Richard" <[EMAIL PROTE
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas calls(busy
or just a hang-up). Is there a way in the dial plan to automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure e
> > I am trying to construct a macro for long distance dialling. I have
> > two internet feeds, I have all routes including Teliax on
> Internet A
> > and a static route to Voxee on Internet B.
Here's an AEL macro I use on our boxes. Modify for your needs.
// dial a number with a range of rout
On Sunday 22 January 2006 14:11, Chris Mason wrote:
> I am trying to construct a macro for long distance dialling. I have two
> internet feeds, I have all routes including Teliax on Internet A and a
> static route to Voxee on Internet B. I thought I could use the dialplan
> entry below which uses t
I am trying to construct a macro for long distance dialling. I have two
internet feeds, I have all routes including Teliax on Internet A and a
static route to Voxee on Internet B. I thought I could use the dialplan
entry below which uses the ChanIsAvail() command to check the
connection, but th
On Mon, 2005-11-14 at 13:11 -0800, Andy Kuo wrote:
> in extensions.conf
>
> exten => _X.,1,Dial(SIP/[EMAIL PROTECTED])
> exten => _X.,2,Dial(SIP/[EMAIL PROTECTED])
>
I dont think that will work quite right starting with BRIStuff. While
congestion() is +1 I believe if the peer is down its +201.
in extensions.conf
exten => _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten => _X.,2,Dial(SIP/[EMAIL PROTECTED])
On 11/11/05, John E. Elkin <[EMAIL PROTECTED]> wrote:
Maybe its already been posted, but i cant find it...
I have an asterisk box running agilevoice (Customer signup and provisioning
Maybe its already been posted, but i cant find
it...
I have an asterisk box running agilevoice (Customer
signup and provisioning system)
I have two sip termination providers.
One provides did and termination. The other provides just my
termination. My big question is.
If the term
> The disk array would be the only expensive add on, more than a normal
> asterisk system. It all depends on how important voicemail is in your
> application, although there are cheaper alternatives (NFS for example,
> but then your NFS server becomes a single point of failure, depending on
> the
On Wed, 2005-04-27 at 11:17 +0300, Zoa wrote:
> Could you explain me some more how i could use dual controllers ? Is
> this done with special harddisks ? What hardware do i need to do this ?
We used a winchester drive array, which is not cheap, and way overkill
for asterisk. EMC makes similar box
Could you explain me some more how i could use dual controllers ? Is
this done with special harddisks ? What hardware do i need to do this ?
/Z.
trixter http://www.0xdecafbad.com wrote:
One thing that could be done is to have a disk array for voicemail and
all with dual controllers. Then plug that
On Wed, 2005-04-27 at 00:52 -0700, trixter http://www.0xdecafbad.com
wrote:
> One thing that could be done is to have a disk array for voicemail and
> all with dual controllers. Then plug that into each of two servers.
> Bind the IP components to a IP that is transportable between machines.
> When
One thing that could be done is to have a disk array for voicemail and
all with dual controllers. Then plug that into each of two servers.
Bind the IP components to a IP that is transportable between machines.
When one fails ifconfig the failover machine to use that IP (could be a
virtual interfac
On 4/26/05, snacktime <[EMAIL PROTECTED]> wrote:
> On 4/26/05, Sean Kennedy <[EMAIL PROTECTED]> wrote:
> > Hi folks,
> >
> > I'm curious; What does everyone do for failover? I have two servers,
> > same os/compilation. I designate one the master, the other the slave,
> > and I rsync the config f
On 4/26/05, Sean Kennedy <[EMAIL PROTECTED]> wrote:
> Hi folks,
>
> I'm curious; What does everyone do for failover? I have two servers,
> same os/compilation. I designate one the master, the other the slave,
> and I rsync the config files once an hour and trigger a restart when
> convenient co
Hi folks,
I'm curious; What does everyone do for failover? I have two servers,
same os/compilation. I designate one the master, the other the slave,
and I rsync the config files once an hour and trigger a restart when
convenient command on the console. These two servers are setup in the
dns
buy 2 load balancer to failover between themselves.
Best Regards
Matt
- Original Message -
From: "Mitchel Constantin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, March 29, 2005 11:34 PM
Subject: Re:
On Wed, 30 Mar 2005 05:03:33 +0800, El Flynn <[EMAIL PROTECTED]>
wrote:
Rich Adamson wrote:
No, that's a service, or at least I think it is, the sales garbage
obscures
what it really is so who knows.
What I need is a little box that diverts calls if the PBX goes down.
FYI, the topic has bee
On 23:34, Tue 29 Mar 05, Mitchel Constantin wrote:
> Matt,
>
> This isn't meant as a flame, rather I'm curious about what other
> people think about the following situation...maybe it's just the
> philosopher in me, what happens when the load balancer fails?
>
Good point. Was thinking the same t
<[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, March 29, 2005 7:11 AM
> Subject: RE: [Asterisk-Users] Fail over
>
> > > No, that's a service, or at least I think it is, the sales garbage
> obscures
>
List - Non-Commercial Discussion"
Sent: Tuesday, March 29, 2005 7:11 AM
Subject: RE: [Asterisk-Users] Fail over
> > No, that's a service, or at least I think it is, the sales garbage
obscures
> > what it really is so who knows.
> >
> > What I need is a little bo
> Some of their products are programmable too, where you can
> send TCP messages to
> initiate the switching process. Check out their website for
> more products.
>
That's perfect, because I use a Nagios monitoring system that can tell if
the Asterisk system is running and tell the fail-over
Rich Adamson wrote:
No, that's a service, or at least I think it is, the sales garbage obscures
what it really is so who knows.
What I need is a little box that diverts calls if the PBX goes down.
FYI, the topic has been discussed previously on the list, and the
problem that you're trying to addre
> No, that's a service, or at least I think it is, the sales garbage obscures
> what it really is so who knows.
>
> What I need is a little box that diverts calls if the PBX goes down.
FYI, the topic has been discussed previously on the list, and the
problem that you're trying to address is far
On Tue, 29 Mar 2005 09:40:08 -0400, Chris Mason <[EMAIL PROTECTED]> wrote:
> No, that's a service, or at least I think it is, the sales garbage obscures
> what it really is so who knows.
>
> What I need is a little box that diverts calls if the PBX goes down.
>
The Sipura 3000 does this. That i
15 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Fail over
>
>
> There's many solutions.. One being www.voiceguard.com I think
> might be what
> you want.
>
> - Original Message -
> From: "Chris Mason
There's many solutions.. One being www.voiceguard.com I think might be what
you want.
- Original Message -
From: "Chris Mason" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, March 29, 2005 8:01 AM
Subject: [Asterisk-Users] Fail over
For all my PBX installations I want
For all my PBX installations I want to have Fail Over on the main incoming
PSTN line so that a power outage does not leave the offices stranded. Is
there any commercial solution to this? I would rather a finished product
than a home soldering project.
Chris Mason
[EMAIL PROTECTED]
Box 340, The Val
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