Displaying 20 results from an estimated 3000 matches similar to: "queue behaviour"
2006 Feb 22
0
R: queue behaviour
That's exactly what I was looking for.
By the way, I discovered Local channels to fork into dialplan.
I also discovered that roundrobin policy does not work as I expected, but that's another story.
Thanks for help,
_fangi_
-----Messaggio originale-----
Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Chris Bagnall
Inviato: luned?
2005 Oct 14
1
Incoming call problem - ringing SIP devices report busy
Hi all,
I have 12 SIP phones at a particular site all connected to a local asterisk
server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming
calls. An IAX gateway is used for outbound calls. At the moment, when an
incoming call comes in, asterisk dials every SIP phone like so:
Dial (SIP/1&SIP/2&etc.)
This has worked fine for some months, but I noticed a few days ago
2006 Feb 24
2
Asterisk Topology
Hi List,
Im planning on setting up asterisk for a large scale enviorment, with
multiple sites.
We will be doing quite a bit of inner office calling at each site, and want
to place a smaller scale * box at each site with no PRI's, and have that
connect to our main * servers at our data center that will have the PRI
connections.
Can this be done? I havent seen to much of this on the mailing
2007 May 08
3
Vista compatibilty in SIP softphones
Greetings list,
I've noticed over the last couple of weeks that, unsurprisingly, nearly every new PC seems to be coming with Vista these days. I expect it'll only be a matter of time for all of us before clients start needing Vista-compatible softphones (if it's not already happened).
So, what's the story with Vista compatibility amongst the softphones currently out there?
2007 Nov 28
3
Asterisk on multi-homed systems
Greetings list,
I remember a discussion many months ago which ISTR concluded that asterisk didn't play nicely at all in multi-homed setups (e.g. SIP packets not being sent out through the same interface they were received on, etc.).
Is this still the case, or are there versions which have resolved the issue? Even if it's still the case, is this only a problem for SIP, or does it affect
2007 Mar 22
2
Linksys/Sipura SPA-942 phones in larger deployments
Greetings list,
Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with
2008 Jan 22
3
Voicemail - is it possible to automatically use the extension being dialed from?
Hi,
Is it possible to dial voicemail from a particular phone line and
automatically enter the extension that is being dialed from, thereby only
prompting for the password?
I've been searching around to find if this is possible, but I haven't been
able to find an example of this. I have a feeling it's more of a endpoint
function, but I thought I'd ask if anyone has accomplished
2007 Sep 25
2
Point-to-Point SIP link without registration
Greetings list,
I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls.
One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either
2008 Jan 16
4
Unable to open master device '/dev/zap/ctl'
Hi,
I'm using zaptel-1.2.22.1 with asterisk-1.2.10 and following steps to
make zaptel working...
OS is gentoo linux 2006.1
Hardware:
---------
0000:05:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
Subsystem: Unknown device 8085:0003
Flags: bus master, medium devsel, latency 32, IRQ 22
I/O ports at b400
Memory at ff900000
2007 May 11
2
Dundi and unknown remote peers
Hi guys,
Is it possible to allow remote peers to connect to your local DUNDi
Asterisk box, even if you don't have them listed in the dundi.conf?
Alex
2006 Feb 27
2
courtesy message calling mobile phones
Hi everybody.
Just noticed that when calling a mobile phone, Asterisk doesn't bridge
the voice message by telco if mobile is unreachable, but keeps on
ringing till it receives a hangup signal. I think this is due to the
fact that the message is played without the call has been answered, but
I'm wondering if there's some way to let Asterisk realize it. All I see
in the CLI is the line
2008 Mar 08
1
PRI suppliers in Switzerland
Greetings list,
I posted this to the -biz list a few days ago. In hindsight, I think it would have been more appropriate posted here, so apologies to those on both lists who've now seen this twice.
I have had a request to provide 2x PRIs to a site in Lausanne, Switzerland, but my knowledge of the Swiss Telco market is non-existent.
Are there any folks on the list who've experience in
2008 Apr 02
1
CentPBX mirror?
Greetings list,
Not exclusively asterisk-related, but I've noticed the CentPBX site has been offline the last few days. Anyone know the reasoning behind that, and more importantly, is anyone mirroring it?
Thanks in advance.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons
2007 Apr 19
2
extensions.conf #include behaviour
Greetings list,
A quick question regarding extensions.conf #include behaviour if I may. I'm sure someone will know the answer off the top of their head...
How does asterisk handle "overloading" of contexts. For example, say an extension exists in extensions.conf as follows:
[incoming]
<some stuff>
Then one includes a, b and c.conf, each of which also contains:
[incoming]
2007 May 03
1
Connections rejected in DUNDi requests
Greetings list,
Wondering if anyone's come across this before.
I've configured a couple of our servers with a "privatedundi" context to allow calls to still flow between extensions even if they're registered to different servers . The DUNDi lookups seem to work fine, evidenced by the following on the originating server:
-- Called
2020 May 01
4
Length of dial string
Hi all
as per the new release notice for 13.33.0 received today - can anyone advise
me the max limit of the string to the Dial Command - see
* [ASTERISK-27946
<BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] -
dial (API): Storage of dialed target uses AST_MAX_EXTENSION
when it shouldn't
I have been fighting with this issue for months trying to find a solution I
2020 May 01
1
Length of dial string
Hi Dovid
Yes was one of the options but as the required list is dynamic becomes very
messy - and all combinations problem - where as "call all workers job xxx"
is what is needed so the ability to call 20+ numbers is what is needed - agi
does a database search for all jobx workers and constructs a dialstring with
SIP, DAHDI and Local devices.
Can someone tell me where to find maximum
2006 Jun 27
2
7960 help: transferring calls
Greetings all,
Not specifically an asterisk query, but a couple of transfer queries that
I'm sure are obvious to folks who use these phones all the time:
1) how does one do a blind transfer? When a call is answered and one hits
the transfer button, followed by an extension, one has to wait for the other
party to answer, then hit transfer again, before the call is released. I'm
sure there
2007 May 23
16
WiFi SIP phones
Greetings list,
What are people's experiences with WiFi SIP phones?
When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices.
I assume things must have moved on somewhat since
2006 Jan 05
5
OT: SIP aware firewalls?
Hi All,
Until now I've only used IAX2 to connect to ITSPs. I've been toying
with a SIP connection to Gizmo Project, but not yet successfully. It
brings to mind a question. At what point does it make sense to consider
a SIP-aware firewall such as those from Ingate?
I'd hate to move away from my m0n0wall, which is open source, easy to
manage and has served me brilliantly for two