oks the Asterisk doesn't know about the call-id. Why not?
Doug.
> -Original Message-
> From: Douglas Garstang
> Sent: Tuesday, March 28, 2006 8:30 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: NATted phones transferring calls
> -Original Message-
> From: Alexander Lopez [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, March 28, 2006 9:04 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] NATted phones transferring calls -
> BUG0003710
>
>
> I wont go into the details of
Do a trace with ngrep or ethereal and see if the phone is sending back a sip
DECLINE or similar. If it is, I'd say you have an Asterisk bug.
Doug.
> -Original Message-
> From: Stephen Kratzer [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, March 28, 2006 7:51 AM
> To: asterisk-users@lists.dig
I made a call from 3254102 to 2944093. I then tried to do a transfer to
3254107.
IP addresses have been changed to protect the innocent.
It appears this related to bug 3710. It's unclear from the bug if the problem
has been fixed or not. If it hasn't, then this seems pretty serious and would I
I just realised my problem seems to be related to bug 0003710 - "0003710:
[patch] Consultative transfers between asterisk servers". It's unclear from the
bug info if this problem has been resolved yet. Anyone know?
Doug.
> -Original Message-----
> From: Douglas Gar
I made a call from 3254102 to 2944093. I then tried to do a transfer to
3254107.
IP addresses have been changed to protect the innocent.
Here's the REFER that the phone at 2944093 sends directly to Asterisk:
U 216.186.128.68:5060 -> 216.186.142.203:5060
REFER sip:[EMAIL PROTECTED] SIP/2.0.
Via:
How
does Fast AGI help?
-Original Message-From: Giovanni Miano
[mailto:[EMAIL PROTECTED]Sent: Monday, March 27, 2006 1:00
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Config File
ManagementYou can use FastAGISee http://www.ast
I have the following python AGI script.
I know it's been abstracted, but it's still pretty easy to
see what's happening.
self.agi.channelAnswer()
self.agi.wait(1)
self.agi.execCmd("background","enter-conf-call-number","")
self.agi.execCmd("Read","confNum|||","")
I'm curious (ok, well I admit it - it's for perosnal gain) what methods people
are using to manage asterisk config files when they have multiple asterisk
systems?
Some sort of revision control such as cvs,rcs or subversion?
A central 'config server' where you edit the files and then rsync them
Easy to configure, lots of options and features, excellent quality speaker for
hands free. Although the 301 is nothing to get excited about. The 501 and 601
are much better.
Doug.
-Original Message-
From: Nick Hoffman [mailto:[EMAIL PROTECTED]
Sent: Sun 3/26/20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Copying SIP Subscriptions
On 3/26/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> I'm pretty sure I already know the answer to this, bu
I'm pretty sure I already know the answer to this, but...
Is there a way to copy/transfer/replicate sip subscriptions from one asterisk
system to another, for the purposes of HA? You coudln't even write a script to
do it I don't think. You can do an 'asterisk -rx sip show subscriptions' but
th
know the username so it knows what entry in
> iax.conf to use for that information, such as the key to use.
>
> Joshua Colp
>
> - Original Message -
> From: Douglas Garstang
> [mailto:[EMAIL PROTECTED]
> To: Asterisk Users Mailing List -
> Non-Commercial Discu
Well,
actually I did... sort of... I picked a random post and hit reply as I normally
do. I forgot to clear the old text, but it's obvious that in no way it detracted
from the other person's post, and the use of the word 'hijack' is highly
dubious..
Doug.
-Original Message-From:
gt; IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]
>
> Joshua Colp
>
> - Original Message -
> From: Douglas Garstang
> [mailto:[EMAIL PROTECTED]
> To: Asterisk Users Mailing List -
> Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
> Sent:
> Sat, 25 Mar 2006 14:55:
- Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] RE: IAX Incoming/Outgoing
>
>
> Douglas Garstang wrote:
> > This is INSANE! My calling system has this iax.conf:
> >
>
> Search the archives for mails about separating
> originations/terminations
> by re
o send a username of
PBX3???
-Original Message-----From: Douglas Garstang
Sent: Saturday, March 25, 2006 11:16 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] RE: IAX Incoming/Outgoing
Well, right now I have this on box1:
[p
Actually, I commented out [pbx3] on the caller, and the callee is STILL
seeing pbx3 as the username. That's even more bizarre. I am sendng pbx1 as the
key from the pbx1 system, and pbx2 is matching it against pbx3???
Huh???
-Original Message-From: Douglas Garstang
.
From: Douglas
Garstang [mailto:[EMAIL PROTECTED] Sent: Saturday, March 25, 2006 12:37
PMTo: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: [Asterisk-Users] RE: IAX
Incoming/Outgoing
12 hours later...
still playing with this. Anyone got any
12
hours later... still playing with this. Anyone got any
ideas?
Doug.
-Original Message-From: Douglas Garstang
Sent: Friday, March 24, 2006 10:53 PMTo: Asterisk Users
Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List -
Non-Commercial
I'vce got three Asterisk systems here that I'd like to be able to place calls
between with IAX. As usual, I've spent several hours playing with it, really
getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2,
pbx3, should be able to connect to every other. Do I need separat
cution at the current
> extension if the
> destination channel hangs up.
>
> Aaron
>
> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>
> > I just changed the macro to:
> >
> > exten => s,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],25,wW)
> >
extension
> that handles a different DIALSTATUS than ANSWER to cover good calls.
>
> Aaron
>
> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>
> > Nope. Still no go.
> >
> > Caller has this:
> >-- Hungup 'IAX2/acdserver1-2'
> > == S
ted non-zero on
'SIP/2944093-9ef2' in macro 'DialIAX'
== Spawn extension (macro-DialIAX, s, 1) exited non-zero on 'SIP/2944093-9ef2'
> -Original Message-
> From: Douglas Garstang
> Sent: Friday, March 24, 2006 4:50 PM
> To: 'Asterisk Users Mail
>
>
> Why do you have s-ANSWER jumping to s-OK? Try putting a NoOp
> in s-ANSWER
> and see if it's making it there... Also, when the call
> doesn't make it
> through, does it jump through the s-DIALSTATUS priorities?
>
> Aaron
>
> On Fri, 24 Mar
>
>
> Looking at your macro, I don't have any MacroExits in mine.
> I use AEL,
> and it doesn't put that on the macros. Try changing your
> MacroExit to a
> NoOp(Macro Finished) and see if that drops you back into the
> original call
> structure.
>
ing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Transferring a call with IAX
>
>
> Hmm... and nothing in the macro after the dial command is
> being executed?
> What does the CLI say on the caller server when the ACD server is
> finished?
>
> A
> Change priority 4 on the ACD server to a Hangup and ignore
> what I said
> before about putting in priority 5. Put the macro call you
> had on the ACD
> server on the PBX server, and that should fix your problem.
> Since you're
> having the ACD server do a macro of
e" server, and then from there to another server for
> voicemail?
>
> Aaron
>
> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>
> > Thanks Aaron, but nope... that didn't do it. I put an
> explicit hangup right after the Queue app on the ACD server,
> and I see thi
isn't hanging
> up the line
> when it runs out of the queue. Add this, and it should work for you:
>
> exten => oe_custcare,5,Hangup
>
> Let me know if that works :)
>
> Aaron
>
> P.S. It's the same on both servers, just the server names are
> swit
I'm just starting testing on, but works like
> a charm with
> the Cisco's. This gives us the redundancy we need with the
> system, and
> hopefully the Polycoms handle it well.
>
> Aaron
>
> On Fri, 24 Mar 2006, Douglas Garstang wrote:
>
> > Don&#
t; original server continues in it's own call pattern until it
> hangs up or
> the user hangs up. This way, whatever server the call comes
> from handles
> all call control except the actual conversation when needed.
>
> Aaron
>
> On Fri, 24 Mar 2006, Douglas Gars
in, I'm kinda excited to see how well they
> handle the
> redundancy. The way we have it set up, we can drop a whole
> new server
> into the cluster, change like 3 lines of dialplan code, and
> add the server
> to the proxy list and voila, another server handling more
Here's an interesting question:
If I transfer a call from Asterisk system to another with IAX, is there any way
I can get control back on the original system? Or.. do I lose control, and the
dialplan has to continue on the new system?
Scenario is we transfer calls to an Asterisk system that han
Here's the $64,000 question... how are you handling redundant registrations?
> -Original Message-
> From: Aaron Daniel [mailto:[EMAIL PROTECTED]
> Sent: Friday, March 24, 2006 11:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Asterisk Fa
In short, does this work yet?
ie putting agents into Realtime. Can't find any info on it...
Thanks,
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.dig
Do queue names have a max length? I have a queue named 'oneeighty_techsupp' in
queues.conf, and a 'show queues' returns a truncated queue name. Is that just a
display bug, or do queues names have a max length of 12?
demeter*CLI> show queues
oneeighty_te has 0 calls (max unlimited) in 'rrmemory'
ueues
Let's try the obvious first, how about
exten => q_main,1,Answer
exten => q_main,2,etc.
Fingers crossed :)
On 3/24/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
Here you go
I have found, however that not using the agent time-out is bad because
the queue continues to send a caller to the same agent, even though that
agent isn't at his/her desk to answer the call. Automatically logging
that agent out allows the call to be passed to the nex
Here you go. I'm not sure this is much use. It's a bit hard to explain as I
have one system calling another via IAX where the Queue() command is
executed... Calls are VOIP->VOIP, on our network... This case below is where
each agent (there's 6) is rung for 30sec, but the Queue aborts after 120s,
If you referring to me Aaron, I don't recollect every using the word 'stupid'.
I have said many times that the way certain things are done for an Enterprise
Class piece of software are unacceptable, or maybe reacted in incredulation
that it would even be considered being done in a certain way.
erisk. Hell, some of us have even helped out a little
>with the development.
Which rude remarks towards developers are you specifically referring to? I
don't seem to recollect any. I have actually received a number of personal
emails from people who completely understand where I am coming from and
I don't see why I'd be getting CHANUNAVAIL. As I said, all the agents are
logged in and are in astdb and 'sip show peers'.
I just had another go, I passed a timeout of 300 to the Queue() command and set
timeout=30 in queues.conf for the queue. The Queue timed out after TWO minutes
(120s) and o
Thanks, but I don't really follow your reply. Agents are being logged in with
AgentCallBacklogin. If I do a 'show agents' command, I see them... all nice and
logged into their queues...
-Original Message-
From: CC Jay [mailto:[EMAIL PROTECTED]
Sent: Thu 3/23/200
Rather than continue to go in circles with queues, I'll ask upfront.
What exactly does the timeout option in queues.conf specify? The docs
contradict themselves on the meaning. What exactly does the timeout option
passed to Queue()? Docs are flaky there also.
I have a BIZARRE situation where I'
ain,1,Queue(oneeighty_main300)
...
[oneeighty_main]
musiconhold = default
timeout = 15
...
Cheers.
On 3/24/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
Egads. Getti
ork because of this :(.
>
>
>
> > -Original Message-
> > From: BJ Weschke [mailto:[EMAIL PROTECTED]
> > Sent: Thursday, March 23, 2006 2:00 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Re: Fw: anyb
Egads. Getting queues to work is like pulling teeth.
extensions.conf:
exten => q_main,1,Queue(oneeighty_main1)
exten => 80014055,1,Dial(SIP/80014018,15,tr)
exten => 80014057,1,Dial(SIP/80014018,15,tr)
exten => 80014052,1,Dial(SIP/80014018,15,tr)
queues.conf:
[oneeighty_main]
musiconhold = def
it clears the settings on a single phone,
> leaving all
> the others untouched. We do it all the time when we
> reconfigure a phone.
>
> Aaron :)
>
> On Thu, 23 Mar 2006, Douglas Garstang wrote:
>
> > Ok Andrew. Here's one for you... I just changed qua
How can a reload clear registrations?
If I 'reload' without using realtime, I keep my sip peers (as well as astdb). I
can still contact other phones... registration info is still there.
If I `reload` with realtime, I lose my sip peers (but astdb remains). I can
_STILL_ contact other phones
*lol* I think he was referring to a ASTERISK reboot, not a phone reboot.
> -Original Message-
> From: Olle E Johansson [mailto:[EMAIL PROTECTED]
> Sent: Thursday, March 23, 2006 1:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Re: Subsc
I also just changed the callerid column for a use in realtime. Doing a
'realtime load sippeers name 80014052' shows the OLD value for callerid. Do I
have to do a reload here? If I do, I'll lose all my sip peer information - BAD!
> -Original Message-----
> From: Dou
e E Johansson [mailto:[EMAIL PROTECTED]
> Sent: Thursday, March 23, 2006 11:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Ok... what is 'sip show peers'
> really used
> for?
>
>
>
> 23 mar 2006 kl. 19.16 skre
My bad. Ignore this reply.
> -Original Message-
> From: Douglas Garstang
> Sent: Thursday, March 23, 2006 12:12 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Ok... what is 'sip show peers'
> really used
&
I _think_ you can do a 'sip reload' instead of a 'reload' and keep your BLF...
But as for a server boot don't be crazy! According to people on this group,
that's just plain never required in the real world.
> -Original Message-
> From: mustardman29 [mailto:[EMAIL PROTECTED]
> Sent: T
lsmith [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, March 22, 2006 10:46 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
>
>
> On Wednesday 22 March 2006 11:34, Douglas Garstang wrote:
> > First thing that com
for?
>
>
>
> 23 mar 2006 kl. 19.16 skrev Douglas Garstang:
>
> > I'd love to understand what the function of the peer list returned
> > by 'sip show peers' is for, especially when Realtime is used.
> >
> > If I start Asterisk with realtim
s you
> what IP address Asterisk knows for a peer, as well as the qualify
> status.
>
> On 3/23/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> > I'd love to understand what the function of the peer list
> returned by 'sip show peers' is for, especially
I'd love to understand what the function of the peer list returned by 'sip show
peers' is for, especially when Realtime is used.
If I start Asterisk with realtime enabled, a 'sip show peers' yields none. As
each peer (phone) registers, or a call is made to the peer, Asterisk adds them
to the l
gt;
> What does your dialplan look like?
>
> On Wed, 22 Mar 2006, Douglas Garstang wrote:
>
> > Arrgh.
> >
> > I just made a call with Asterisk to extension 2944093. That
> extension exists in astdb and I have rtcachefriends=yes in
> sip.conf. Asterisk di
Sounds like an implementation bug. I have not seen this issue with Polycom 601
phones and Asterisk queues. I'd suggest you get ngrep or tethereal and run it
on your Asterisk system (port 5060) to see who's sending what to where.
Doug
-Original Message-
From: [EMAIL P
Arrgh.
I just made a call with Asterisk to extension 2944093. That extension exists in
astdb and I have rtcachefriends=yes in sip.conf. Asterisk did a database
query...
SELECT * FROM ast_sip_users WHERE name = '2944093'
Uhm... Why?
Doug
___
--Bandwi
> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, March 22, 2006 8:55 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
>
>
> On Tuesday 21 March 2006 18:12
> -Original Message-
> From: Olle E Johansson [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, March 22, 2006 8:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Realtime / SIP Peers etc
>
>
>
> 22 mar 2006 kl. 15.02 skrev Andrew Kohlsmith:
risk-Users] Programming the Manager API
Douglas Garstang wrote:
> Yikes. Java. Yuck. I'll stick with Python... Thanks anyway.
> I just worked it out... you can supply an actionid to the request to
know what reply to look for, although it will stil
ailto:[EMAIL PROTECTED]
Sent: Tue 3/21/2006 10:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] Programming the Manager API
Douglas Garstang wrote:
> I just started poking aro
I just started poking around with writing a python module to interface to the
Manager API, and it suddenly hit me... how the heck are you supposed to program
this thing?
All the events seem to be dumped to all the open connections. If I send a
command, such as a login, there seems to me to be
I've got 1.2.5...
-Original Message-
From: JR Richardson [mailto:[EMAIL PROTECTED]
Sent: Tue 3/21/2006 8:19 PM
To: asterisk-users@lists.digium.com
Cc:
Subject: [Asterisk-Users] RE: VoiceMailMain(@context) Problem with
Option 5(Advanced)
a given.
Doug.
> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, March 21, 2006 5:04 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
>
>
> On Tuesday 21 March 2006 18:1
Oooo I think I am gonna poo my pants.
Using the microbrowser on a Polycom 601, I was able to get it to execute a cgi
script upon selection of an item. The cgi script used Net::Telnet connect to
the manager interface on another Asterisk system, call the user back at their
phone and then bridge t
I had to drop realtime with sip users. If you do a reload or a restart, you
lose all the sip peer information (even with rtcachefriends=yes). That just
wasn't acceptable for us.
> -Original Message-
> From: JR Richardson [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, March 21, 2006 4:06 PM
I had the same problem yesterday. I thought it might have been a realtime
problem. Guess not.
Bloody annoying too.
> -Original Message-
> From: JR Richardson [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, March 21, 2006 2:52 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users]
- Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Native MOH - Convert mp3 to ulaw
>
>
> Douglas Garstang wrote:
> > I tried that earlier today... found it somewhere online...
> This is what I get...
> >
> > [EMAIL PROTECTED] mp3]# sox -V fpm-calm-river.m
Convert mp3 to ulaw
>
>
> Douglas Garstang wrote:
> > Tried it. It doesn't seem to like converting to ulaw,
> unless I'm doing something wrong. The man page mentions ulaw
> briefly, but doesn't say HOW to convert to it.
> >
> > Doug.
> >
&
> Bob McDowell
>
>
> -Original Message-----
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Douglas
> Garstang
> Sent: Tuesday, March 21, 2006 11:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users
I have rtcachefriends=yes in sip.conf.
It is caching friends because as I said in my post, astdb has all the contacts,
ie they're cached.
It's the behaviour of 'sip show peers' that's not working.
> -Original Message-
> From: David Thomas [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, March 2
ercial Discussion
> Subject: Re: [Asterisk-Users] Realtime SIP Persistency
>
>
> Douglas Garstang wrote:
> > I've been using realtime for sip users information.
> >
> > I noticed that when you are doing this, if you do a
> 'reload' or restart asterisk, t
I've been using realtime for sip users information.
I noticed that when you are doing this, if you do a 'reload' or restart
asterisk, the information in a 'sip show peers' goes away. When I do this, MWI
stops working. I always though MWI used the astdb file ('database show') to
determine where
Ready
to scream here..
1.
After 6 months with Asterisk I'm STILL trying to understand the difference
between a SIP user, friend and peer.
2.
Exactly what resource does Asterisk use to send MWI to registered phones? I
thought it was astdb?
3. It
looks like it isn't astdb. It looks like i
I'd like to use native moh instead of with mpg123... for some reason the
processes never bloody die.
For native moh to not spawn an external player, I'd need to convert the default
supplied moh sound files in /var/lib/asterisk/mohmp3 to ulaw and g729 format.
Anyone know of a free, easy way to c
Ugh.
I have voicemail set up for realtime...
mysql> SELECT * FROM ast_vm_users;
+--+-+---+-+--+--+---+---+-+
| uniqueid | customer_id | context | mailbox | passwo
Well, this is a major pain in the ass.
I got realtime static working for sip.conf. 'Great!' I thought. That was until
I realised I couldn't use it.
Our Asterisk systems are using OSPF and listen on interface lo:1. Asterisk
doesn't like to use an interface name for it's bindaddr setting, so you
undant.
Aaron
On Mar 16, 2006, at 1:03 PM, Douglas Garstang wrote:
> I know someone who's at VON this week. Apparently Mark Spencer was
> up there talking about how Asterisk supports SRV. Sounds like
> vaporware to me
come from? Will this even work? Would I be so
frustrated if this stuff was documented somewhere?
> -----Original Message-
> From: Douglas Garstang
> Sent: Saturday, March 18, 2006 11:55 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users
Just spent hours dicking around with SIP Realtime.
Every time a phone came up and sent a registration to Asterisk, Asterisk would
simply NOT query the database. I had sipusers in extconfig, but added sippeers
as well. NOW I can see Asterisk doing a 'SELECT * FROM sippeers WHERE name =
'2944093'
at 3/18/2006 6:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] SIP Realtime Users
Douglas Garstang wrote:
> Trying to get SIP realtime working here...
>
>
Trying to get SIP realtime working here...
I'm connected to the database...
*CLI> realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username voxadmin for 6 seconds.
I can get information for the extension in question...
*CLI> realtime load sipusers name 2944093
?
>
>
> - Original Message -
> From: "Douglas Garstang" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, March 17, 2006 2:13 PM
> Subject: RE: [Asterisk-Users] Re: regexten
>
>
The polycom doesn't need to re-register... it's already registered on all three
Asterisk systems. :)
> -Original Message-
> From: Gabriel Afana [mailto:[EMAIL PROTECTED]
> Sent: Friday, March 17, 2006 3:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [As
>
> - Original Message -
> From: "Tony Mountifield" <[EMAIL PROTECTED]>
> To:
> Sent: Friday, March 17, 2006 1:04 AM
> Subject: [Asterisk-Users] Re: regexten
>
>
> > In article
> <[EMAIL PROTECTED]>,
> > Douglas Garstang <[EMAIL PRO
Does anyone know how I can run asterisk in the background WITH extra debugging
(ie the -d option) without the console popping up?
Try it... Run 'asterisk -d' and you'll get the console.
Huh?
Maybe it doesn't make sense to run in the background with debugging turned up,
but I want to see the extr
Trying to find a solution to a sticky problem here.
We have 3 OpenSER systems. Phones register with the OpenSER systems, and after
they authenticate the user, pass the registration info using OpenSER's send()
command to all Asterisk boxes sitting behind them. Each asterisk system then
knows abo
le work.
Doug.
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Thu 3/16/2006 11:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] regexten
On 17
Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] regexten
On Thu, Mar 16, 2006 at 04:41:42PM -0700, Douglas Garstang wrote:
>
>
> > -Original Message-
> > From: Luigi Rizzo [mailto:[EMAIL PROTECTED
n
Cc:
Subject: Re: [Asterisk-Users] Feedback from VON expo! Info on * HA
andPolycomphone!!
On 17/03/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> David,
>
> How's DUNDi make this redundant? The way I unde
Gabe,
I'm the guy with the three SER boxes. All three SER boxes are active. Polycom
phones point to a DNS domain with SRV lookups for all outbound requests. The
SRV queries return the three SER boxes. The three boxes authenticate a user and
then store the user in it's 'location' database befor
Grrr. I'm using outlook web access and there's no way to do inline replies.
Anyway...
Gabriel.
Using SER does not create a single point of failure. You install three SER
boxes. Single point of failure gone.
It does not take several seconds.
If your phones are configured for SRV, and 2/3 of y
David,
How's DUNDi make this redundant? The way I understand it, a phone is only ever
registered to a single Asterisk box at a time. If that Asterisk box where to
fail, callers lose the ability to contact users that where registered on that
box.
-Original Message-
Fro
ing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] regexten
On Thu, Mar 16, 2006 at 04:27:10PM -0700, Douglas Garstang wrote:
> Thanks for the reply, but still no luck.
>
> sip.conf:
> [294409
I spent some time today working on HA option number 1. I don't know if I'd
really call it HA, given that the phones need to have a high registration
frequency to get around a server failing. How often do you have the phones
re-register? Once a minute? Imagine that multiplied by thousands of phon
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