Asterisk with SIP Trunk for 400 concurrent calls

Hi all,

I need to build an Call Center which handles 400 concurrent incoming calls from sip trunk.
How many Asterisk server required to handle 400 Concurrent Calls and what should be the specification of each server?
How can I do the load balancing & what should be the network architecture?

I’ll appreciate if anyone can help me.

Do the both server are Asterisk base? if ok, its better use IAX2 trunk between them to safe your bandwidth.
If no, what is your choose codec? It become more importance(LAN) and depends on rate of your network.
Do you need record all calls? It in important and impact to your server performance.
The callers who want to call with your call center, How they call with you? Do your server is connected to PSTN Network or no?
You must more clear what you need.

There is not an standard Asterisk dimensioning table. But i used for a similar project a server.

Lenovo ThinkServer TD340 Tower Server - 5U, 2-way, Intel Xeon E5-2420 v2 2.20GHz, 32GB Memory

Some tips.

You could use an external hard drive for calls recording.

If you have the necessary bandwidth for handle 400 currents incoming calls try to use wide band codecs. Narrow band codecs use more CPU resources .

implement QoS

Try to avoid transcoding this will release extra charges from your servers.

disable the Re-invite by turning off sendrpid ( Note Re-invites are also used for connected line presentation updates and they are also used for session timers.

You are confusing bitrate with bandwidth. Actually, low bit rate codecs may actually be better if you have no transcoding (call recording, in band DTMF detection, and in band tone signals involve transcoding, not just mismatched codecs), simply because less bytes of data need moving around. Make sure that you have all your voice recordings in all the codecs actually used.

If you have a high number of concurrent calls, and there is nothing that conflicts with direct media, you want re-invites ENabled. COLP and session timer uses of re-invites will not add significant loading - the reason for turning those off is that they can cause interworking problems.

Generally, though, you will not get specific advice on sizing, only statements of what worked in specific cases.

Hi All,

Thank you for your replies.

SIP Trunk is provided by the telecom provider. All calls coming to my PSTN number will be reaching my
Asterisk Server through a SIP Trunk provided by the Telecom provider. We are not recording calls and no DMFT detection.
When a call comes in through the SIP Trunk, we have an IVR which varifies the CallerID with our DB and play a products information voice file and transfer the call back to the SIP Trunk of the provider to connect our Agent who is using a GSM Mobile phone.

I hope you got a clear idea what I want to impelement.

As per my knowledge, I think I should use a SIP Load balancer like OpenSip to and distribute the Calls to the Asterisk Server.

How many Asterisk server required to handle 400 Concurrent Calls and what should be the specification of each server?
How can I do the load balancing & what should be the network architecture?


ITSPs normally can’t cope with direct media, and, in any case, it is irrelevant if you are not connecting calls through.

They may constrain your choice of codecs.

The main thing to remember in this configuration is to record the announcements using the ITSP’s choice of codecs, so that they can be played down the wire with no transcoding overhead…

It is unlikely that you will get anyone to commit to a direct answer to the number of servers.

In terms of a load balance, a quick web search suggest you may want to look at … tcher.html


Thanks for your reply.

By Default my telecom provider is giving SIP Trunk with G.711 codec . I can use Voice Compression card for using G.729a or G.723.1

I’m confused for the Network architecture with load balancing and Server configuration.