[Asterisk-Users] Asterisk Hardware Recommendation
mattf
mattf at vicimarketing.com
Fri Apr 29 07:21:00 MST 2005
If price would truly not an option just get one of the Signate Telephony
5000 servers(http://www.signate.com/pbx.php) They are about $18,000 and
allow you to have upto 5000 SIP streams go through it. You could have that
be your gateway and do the SIP->IAX through that machine and scale upto 100
T1s if you want.
But that is a bit steep. So on to your choices. I would really say that the
setup you choose will depend on what kind of users you have as well as how
often you need to change/add users to the system and how the users are using
the system at what times. Any of them that you listed could work depending
on how they are used, but in some cases you may not want to use some of the
scenarios listed because they would either be incapable of meeting your
needs or overly complex to manage.
The easiest and cheapest one would actually not be listed:
Scenario 6:
Direct SIP->Zap on two separate servers half SIP users on each server
PSTN <--2xT1--> A1 <----> SIP_Agents
PSTN <--2xT1--> A2 <----> SIP_Agents
There is really no reason to have another 2 servers running IAX to the T1
servers, and this is simple and easy to set up and involves only 2 machines.
The next setup I would recommend would be Scenario 4, although you will have
to get a machine with a fast/wide BUS(like an Apple G5) to handle ever
increasing numbers of SIP->IAX streams as the system would grow.
If you can explain more about what kind of use this system will have I can
give a better recommendation.
MATT---
-----Original Message-----
From: Daniel Salama [mailto:dsalama at user.net]
Sent: Thursday, April 28, 2005 10:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation
This is great information. I have the following questions based on a
hypothetical scenario and some assumptions:
Based on the price of these configurations, I wouldn't even mind
putting two servers each with 2 T1s just so that I could get all calls
recorded and distribute the risk of failure.
Now, I don't know if it would make a difference or not, but here it
goes:
Assuming the cost of the systems is of no importance for a moment
(actually looking for the most scalable and reliable solution), which
would be a better approach to solve the issue of activating 4 T1s which
will be constantly taxed with load and be able to record all
conversations:
Scenario 1: 4 T1s into Asterisk (A1) where all SIP agents register.
Call recording in A1.
PSTN <--4xT1--> A1 <----> SIP_Agents
Scenario 2: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding.
Asterisk (A1) connects to Asterisk (A2) via IAX where all SIP agents
register (IAX to SIP transcoding). Call recording in A1 or A2.
PSTN <--4xT1--> A1 <----> A2 <----> SIP_Agents
Scenario 3: 4 T1s into Asterisk (A1) to do TDM-IAX transcoding.
Asterisk (A1) connects to Asterisk (A2) via IAX where half of SIP
agents register to, and the other half would register in A1. Call
recording in A1 and/or A2.
PSTN <--4xT1--> A1 <----> SIP_Agents
A1 <--IAX--> A2 <----> SIP_Agents
Scenario 4: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX
transcoding. Asterisk (A2) will connect to A1 and A3 via IAX. All SIP
Agents register at A2 (IAX to SIP transcoding). Call recording in [A1
and A3] or A2.
PSTN <--2xT1--> A1 <----> A2 <----> SIP_Agents
PSTN <--2xT1--> A3 <----> A2 <----> SIP_Agents
Scenario 5: 2 T1s into each Asterisk (A1 and A3) to do TDM-IAX
transcoding. Asterisks (A2 and A4) will connect to A1 and A3
respectively via IAX. Half SIP Agents register in A2 and other half in
A4 (IAX to SIP transcoding). Call recording in [A1 and A3] or [A2 and
A4].
PSTN <--2xT1--> A1 <----> A2 <----> SIP_Agents
PSTN <--2xT1--> A3 <----> A4 <----> SIP_Agents
Hopefully you're all able to understand my 5 scenarios. I guess, my
questions would be:
1) Is there a load limiting factor in terms of whether you do the
Monitor"ing" of the calls when you're doing TDM-IAX transcoding or
IAX-SIP transcoding?
2) Will a single CPU machine handle the 4 T1s to do TDM-IAX
transcoding, if another machine is doing the actual recording (IAX-SIP
transconding) (Scenarios 2,3,4,5). Basically, just setup "cheap"
Asterisk boxes to act as VoIP gateways and the distribute the "load"
and/or intelligence on other Asterisk boxes to connect SIP agents and
all dialing rules, etc?
Thanks,
Daniel
On Apr 28, 2005, at 9:17 PM, mattf wrote:
> You can throw together a single P4 3GHz with 1GB RAM and 2 x 80GB SATA
> HD
> for about $600. One of those can easily handle a Sangoma dual T1
> card($900)
> or a Digium quad T1 card($1400). For that you can have a system for
> about
> $1500-$2000 that will be able to fully record 2 T1s(48 channels) worth
> of
> Zap->SIP conversations. Putting two of those together with a nice big
> fileserver will give you a lot of flexibility, as well as only a
> reduction
> in capacity if one of the servers go down instead of a total outage,
> for
> about the same overall price of a single high-end Dual Xeon server.
> Building
> your system this way from the start will also allow it to scale much
> more
> easily than using just a single very expensive server. You can just add
> another 2 T1s of capacity at any time for just $1500.
>
> I recommend only 50 or less recordings concurrently because that is the
> ceiling that we discovered while trying Zap->SIP recording on both Dual
> Processor server-class systems and single processor cheaper commodity
> computers as well as on SCSI, IDE and SATA drives.
>
> If anyone out the has reliabily done recording of more than 50
> conversations
> I would like to know the hardware architecture of your setup.
>
> Thanks,
>
> MATT---
>
>
> -----Original Message-----
> From: Daniel Salama [mailto:dsalama at user.net]
> Sent: Thursday, April 28, 2005 6:59 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk Hardware Recommendation
>
>
> Thank you again. I will definitely do that. By "cheaper" asterisk
> servers, do you mean single-CPU machines that can handle Quad T1s and
> still do the call monitoring?
>
> BTW, I tried the monitoring without the 'm' option and mounted the
> audio directory via NFS. Big NO NO for everyone. Just do what Matt
> says: copy the -in and -out to archive server separately several times
> a day :) - don't record to NFS mounted drive.
>
> Thanks,
> Daniel
>
> On Apr 28, 2005, at 6:42 PM, mattf wrote:
>
>> I have never been able to do more than 50 concurrent recordings with
>> Zap ->
>> SIP phone calls without the audio skipping and/or breaking up. Also,
>> if you
>> are using Digium TE4XXP and want to do a lot of recording I would
>> recommend
>> against a SCSI RAID card because of the interrupt conflicts that you
>> will
>> run into over time. I would recommend a couple of cheaper Asterisk
>> servers
>> with a dual T1 or Quad T1 board in them and SATA drives, with a nice
>> big
>> archive server that the audio will be copied to several times a day.
>> Also,
>> do not record(Monitor) with the 'm' flag on because this will also
>> lead to
>> more disk read-write while you are already trying to write another 100
>> or so
>> streams. Offload the -in and -out files to the archive server and let
>> it
>> soxmix them together instead. This is the method that we have settled
>> on for
>> our 12 Asterisk servers and it works rather well for us.
>>
>> MATT---
>>
>>
>> -----Original Message-----
>> From: Daniel Salama [mailto:dsalama at user.net]
>> Sent: Thursday, April 28, 2005 5:56 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [Asterisk-Users] Asterisk Hardware Recommendation
>>
>>
>> Hi,
>>
>> I've been reading on the wiki as well as on this list, different
>> suggestions of what to look for when designing an asterisk server with
>> a lot of traffic. By "a lot" of traffic, I mean a box with a a TE4XXP,
>> that will be hit to full capacity (96 simultaneous calls). This box
>> will also deliver these calls to SIP users and record all their
>> conversations via Monitor.
>>
>> I've heard that it's not necessarily a matter of memory (RAM) nor the
>> need to have a multi-processor machine. But what really matters is
>> that
>> the motherboard (architecture) is designed to handle such a high
>> amount
>> of interrupts generated by the TE4XXP, the NIC, the storage array
>> (whether it's SCSI or IDE or SATA).
>>
>> Does anyone have experience with particular brands of either
>> motherboards they recommend are capable to handle this or complete
>> systems (e.g. Dell xxxx or whichever brands), that are ready for this?
>>
>> Thanks,
>> Daniel
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list