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Today's Topics:
1. 5,000 concurrent calls system rollout question (Vic)
2. How to handle "provider UNREACHABLE" in the dialplan?
(Ronald Wiplinger)
3. delaying "answer" for a number of rings or an amount of time
(Brian J. Murrell)
4. RE: 5,000 concurrent calls system rollout question
(Michael Loftis)
5. Re: 5,000 concurrent calls system rollout question
(Michael Loftis)
6. RE: Directed Call Pickup (Alex Barnes)
7. Re: ISDN Eicon Diva Server V-BRI (Bartosz Jozwiak)
8. RE: RE: 5, 000 concurrent calls system rollout question
(William Boehlke)
9. Re: ISDN Eicon Diva Server V-BRI (Jens Vagelpohl)
10. Re: RE: Rewind MusicOnHold? (Dan Journo)
11. Re: ISDN Eicon Diva Server V-BRI (Armin Schindler)
12. Agents, queues and zombies (Steve Rawlings)
13. Fw: Agents, queues and zombies (Steve Rawlings)
14. Re: ISDN Eicon Diva Server V-BRI (Armin Schindler)
15. Re: OT O'Reilly Asterisk TFOT (James Ronald)
16. Re: Blocked Callerid ([EMAIL PROTECTED])
17. Re: fax possibilities ([EMAIL PROTECTED])
18. Re: RE: Rewind MusicOnHold? (Dan Journo)
19. RE: OT O'Reilly Asterisk TFOT (Michael Collins)
20. Re: OT O'Reilly Asterisk TFOT (Mark Phillips)
21. RE: Anyone know a good ITSP in Canada that supports*?
(Technical Support)
22. Slightly OT: OpenPBX.org and Freeswitch (Michael Collins)
------------------------------
Message: 4
Date: Thu, 02 Feb 2006 12:17:03 -0700
From: Michael Loftis <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout
question
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii; format=flowed
--On February 3, 2006 3:56:21 AM +0900 Vic <[EMAIL PROTECTED]> wrote:
Hi, Joash,
thank you for your email. I was very relieved to hear that someone was
already doing this.
Can you please tell me more about your test? Why did you test it in a
first place?
For me, we need to come up with a system that needs to:
1. Handle 5,000 inbound SIP calls
2. offer IVR capability
3. Billing
You'd probably have to do some of your own work on this. * makes 'CDR'
records but...well...you have to be careful how you do your scripts if you
want legible/useable CDRs. There are some apps out there though that will
process and do some sort of billing for CDRs not sure of what where.
I thought that Asterisk would be up to the task, but, I am not sure as
to:
1. How many servers should I consider? 4? 10? Obviously, we will be
talking about probably core Xeon servers if this is what we need.
I'd say atleast 10....maybe more...depending wholly on codec/transcoding
and amount of IVR scripting.
2. How hard would it be to implement?
Well...since your not well versed with *, and you're having trouble
understanding the difference between a protocol and a codec, it might be
really difficult for you. You might want to farm it out. There are a LOT
of * consultancies out there now. If you can get up to speed on asterisk
pretty quickly and the various protocols and codecs then it's not
impossible. The kicker is all the management/maintenance UI's and such.
But you might be able to use something like Signates sigMAN (never used it
or their products).
3. How bad is g729 quality?
4. IVR : if the call is SIP, can we do prompts without transcoding?
You're confusing protocols with codec's here again. SIP is not a codec.
That said if your SIP client is using GSM and there are GSM prompts
available then the asterisk playback functions will use the GSM encoded
prompts.
Earlier you'd mentioned using POTS lines coming in/out. If you're
gatewaying 5k POTS lines you'll need a lot of machines. Because you'll be
doing a lot of transcoding POTS is ulaw or alaw (depending on where in
the
world you are) and unless you use (uncompressed) ulaw or alaw on your SIP
clients (very unlikely scenario) you'll be transcoding to/from GSM. G.729,
or whatever you're using.
Any other suggestions that you might have would really be appreciated.
Joash Herbrink <[EMAIL PROTECTED]> wrote:
I have tested an asterisk server with over 5000 concurrent calls.
The system setup was a P4 HT 3Ghz, 4 Gb RAM, and 1 gbps Ethernet
connection on a cisco 3560 switch.
This works, but puts some serious stresses on the system.
Why don't u considered using g.729 codec, this will at least lower the
bandwidth consumption significantly, and, you can overcome the CPU
resource issue by just using a server grade multi CPU xeon server.
I would never the less still connect the system via 2 ethernet
connections, just for some redundancy, as mentioned before in this
thread.
Bandwidth should be about 24 kbps (half duplex) per call
So, 5000 * 24 is roughly 120 mbps, so a gigabit Ethernet should do just
fine.
Joash
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dustin
Wildes
Sent: Wednesday, February 01, 2006 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout
question
Dinesh Nair wrote:
On 02/01/06 09:29 Damon Estep said the following:
Ok, now lets go for 5000 of them. 160kbps*5000=800000kbps or 800mbps -
full duplex.
Have you ever seen a NIC or switch that can run GigE full duplex at 80%
utilization and not at least start to fall apart?
additionally, 5000 simultaneous SIP calls at 20ms intervals will send,
5,000 * 50 * 2 = 500,000 packets per second (full duplex).
not too many boxes can handle such packet load, in spite of the
relatively small packet sizes.
Why not bond multiple NICs together to do a load balance output? Would
provide redundancy as well.
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--
"Genius might be described as a supreme capacity for getting its
possessors
into trouble of all kinds."
-- Samuel Butler
------------------------------
Message: 5
Date: Thu, 02 Feb 2006 12:18:20 -0700
From: Michael Loftis <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout
question
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii; format=flowed
--On February 3, 2006 4:07:05 AM +0900 Vic <[EMAIL PROTECTED]> wrote:
Hi,
several of your mentioned signant as a viable option.
Has anyone ever used them? Are there any reviews for their products?
Did they just put together a lot of Asterisks into a large scale PC? (I
am still struggling with the concept)
Well I've nebver used it but any single box solution is going to have to
have custom hardware and some custom code in asterisk or asterisk channel
module to run it. A PC can't do echo cancellation on 5k channels. Can't
do codec on 5k channels. It might be able to do (light/simple/short) IVR
on 5k channels though.
Thanks,
Vic
--
"Genius might be described as a supreme capacity for getting its
possessors
into trouble of all kinds."
-- Samuel Butler
------------------------------
Message: 8
Date: Thu, 2 Feb 2006 11:21:08 -0800
From: "William Boehlke" <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] RE: 5, 000 concurrent calls system
rollout question
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
Signate has claimed 5,000 streams, or 2,500 calls, on a single Telephony
Server 5000. The throughput has little to do with Asterisk and a lot to do
with hardware design and operating system tuning. Our very minor code
changes were returned to the project last year.
The benchmark we used to make that initial claim was flawed, however we
have
since replicated the throughput in a different way to save our marketing
bacon.
How we actually achieve the throughput is our intellectual property but we
have a number of customers who are scaling towards and past that traffic
level. One of these days we hope to be able to justify the very large fee
Hammer wants to extract from us to produce a third party verification.
In production environments, of course, systems do more than switch calls.
We
think high volume system design using 32-bit systems of any kind is
complex,
and it's difficult to replicate the volumes without actual customer
traffic
- and by then it's too late. Where do you put voicemail? Where does the
IVR
reside?
When someone needs to switch 5,000 calls with Class 5 services we would
specify a rack of servers. The good news is that it is one rack, not three
of them, but we need more than Asterisk alone, great though it is, to make
everything work.
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Wednesday, February 01, 2006 9:33 PM
To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com
Subject: [Asterisk-Users] RE: 5, 000 concurrent calls system rollout
question
Signate sells a single server that can get you to the call volumes you
need.
Paul Mahler
<mailto:[EMAIL PROTECTED]>[EMAIL PROTECTED]
www.signate.com
[snip]
Past conversations on this topic have generated quite a bit of
controversy within the Asterisk development community, both publicly
here on the list forums as well as in quite a few more quiet
discussions with people who often do not post but have extensive
operational experiences with Asterisk (most of whom monitor the -dev
list and whose replies will be suited to that audience.)
The subject of load on a single chassis is still the most contentious
issue to date. The Signate numbers of >5000 calls per chassis with
RTP are impressive, and there are others who claim more vaguely of
1000, 2000, or more calls into a single P4 server (with or without
media.) Others say that there are inherent limits in the Asterisk
code which prevent more than ~500 calls from being processed with RTP
at any one time. Opterons, FreeBSD, custom Linux loads, Solaris, and
other operating systems or hardware have been offered as the magic
bullets to increase call volumes. Who knows? (1) I will say that
extraordinary claims demand extraordinary evidence, which has been
pretty thin. I believe that most large call processing facilities
still run on distributed systems of some type, as was described in
the primary thread of this discussion on -users. (2)
I know that there are some projects towards testing Asterisk more
rigorously to determine these numbers. However, I would suggest that
the community at large could benefit from a more open examination of
high-end system claims immediately than these (better) long-term
tests which are progressing slowly (if at all.) Let's just look at
the "maximum" numbers. Running a big system? Selling a big system?
Tell us about it, in detail. What are the limits that have been hit?
Be specific. I keep seeing hand-waving, but no programmers have come
forward to say "It won't work because of the way X is implemented in
the file blah.c or libFOO."
To make a bad analogy: I don't want to see the street rods; I just
want to see the top-fuel, rocket-powered dragsters on the line. Any
takers? It sounds like Signate has a contender, but quite a few
people have said that it's impossible without serious modifications
to the code. Others have claimed (publicly or privately) that they
can match those numbers on different hardware.
Here are the criteria:
- Any O/S
- An unmodified version of Asterisk from SVN (or CVS)
OR patches must be available for inspection, as per the GPL
OR you must be a Digium license-holder (patches can be secret)
- All calls are IAX2 or SIP (both in and out)
- No transcoding of any type is required
- All calls are G.711, 20ms OR 30ms packet size
Documentation:
- All O/S documentation, kernel tricks, modules, hacks, patches, or
configuration elements should be documented, but proprietary
information need not be divulged if that is deemed "secret"
- Testing method must be reasonably documented
- Dialplans must be included
- SIP.conf files must be included
- All hardware must be fully described (part numbers required)
TEST #1:
All media must be handled by the server. This is for both legs of
the call. The "canreinvite=no" for SIP and "notransfer=yes" in IAX2
must be set for all calls.
TEST #2:
Media may or may not be handled by the server. Native transfers
should be allowed in both IAX2 and/or SIP.
(1) I have heard various people saying that it is "impossible" for
Asterisk to handle a large number of calls due to architectural
issues (no, it's not just from the people that you'd "expect" to hear
this from.) I've not been able to validate this one way or the other
recently. I am interested to hear what the developer community has
as a comment on this topic. I have an Empirix Hammer system at my
company, but honestly I just don't have the time to set it up to do
testing due to day job time constraints...
(2) There are so many ways to spread calls across an Asterisk array
it makes my head spin, but the question STILL comes down to "how many
calls can a single chassis handle?" Even in a farm of servers, there
has to be a numerator in that ratio.
JT
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