Can We Install 10 PCI Card On Asterisk

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Hi All,

i would like to know if anyone has done or having idea regarding PRI
terminations in Asterisk.

i have a requirement where i need to support 80 PRI in one machine i have found a machine which have 10 PCI slots available

now i am thinking of arranging 8port sangoma card in this pci slots so i can arrenge 10 card in that.

is it possible to run system like that ? is it good idea , can asterisk handle 2400 calls if machine size and RAM is good.

let me know ideas and suggestions.

thanks Dhaval

8 thoughts on - Can We Install 10 PCI Card On Asterisk

  • Last I checked, the highest channel number DAHDI supported was 1023, limiting you to some 34 E1 PRIs.

    We’ve faced stability issues with more than 500 simultaneous calls on a single high-powered server, with no transcoding. However, that’s probably more a limitation of our own architecture and application than a hard Asterisk limit.


    — Raj

  • That sounds like having too many eggs in one basket to me. If you have a problem with that single machine, you’re going to disrupt 2400 calls when you do. A more distributed approach with multiple machines running Asterisk and multiple standalone PRI gateways will be much more fault tolerant if designed correctly.

  • I don’t think Asterisk can handle that many DAHDI channels and I have never heard of an Asterisk box with more that 16 PRI’s.

    Taking a step back, do you really want to put all your eggs in one basket? what if the box fails? That’s 2400 channels going down and unavailable until you fix it. That will cost a log of money and get you angry clients. It makes more sense to spread the lot across different servers. Besides that, is your telco willing to provide you with 80x PRI
    or will they insist on aggregating it to several E3 links or something higher (STM-1)?

    If you really insist on going down this route have a look at FreeSWITCH
    or look at something like Cisco, Alcatel, Telco Systems etc.

    Regards, Patrick

  • Since DAHDI-Linux 2.5.0 there is not a hard coded limit on the number of channels [1] or spans [2] that DAHDI can support. The number of pseudo channels is limited by a module parameter,
    ‘max_pseudo_channels’, which currently defaults to 512 [3].


    I too would be very surprised if DHAVAL is able to run anything close to 80 spans in a single server without modifying the drivers to a) optimize for increased throughput versus minimal latency
    (DAHDI currently favors low latency between bridged channels) and b)
    Rework how conferencing / bridging is done in the drivers so that only channels in conference with one another are checked for audio mixing. I believe the limiting factor would be CPU cycles and not memory.

    As an aside, the most I’ve run in a single server are 24 T1s. But I
    was not optimizing for density either.

    Cheers, Shaun

    Shaun Ruffell Digium, Inc. | Linux Kernel Developer
    445 Jan Davis Drive NW – Huntsville, AL 35806 – USA
    Check us out at: &

  • Your best bet is a carrier class device from someone like Adtran and convert the PRIs to SIP before passing the calls to Asterisk.

    —–Original Message—

  • Hey All,

    Thanks for everyone input on this, this was just mine thoughts to put 80
    PRI line in that.but after reading inputs from everyone i think there are some options to achieve it.

    it means i need to put a gateway which convert my SIP calls to PRI line and another options is to put multiple asterisk boxes and each box have maximum 16 pri lines . now which is best choice to work on further. also i need to consider hardware sizing too as if gateway is expensive i would go with pri cards. also if i choose gateway then also i need to put multiple asterisk boxes.

    let me know your thoughts.

    thanks Dhaval

  • FWIW…

    Our largest setup consists of some 2000 SIP users distributed over 2
    boxes (sip concentrators). The PSTN interface is through another set of boxes with up to 24 PRIs per box (dial banks).

    Users log into one of the sip concentrators with soft (Qutecom) or hard SIP phones. When they place a call, it’s automatically distributed to a PRI on one of the dial banks. The PRI selection is weighted random, with individual preference sets being assigned for each group of callers.

    The biggest issue we faced was figuring out that you can’t have more than one PSTN provider on a single dial bank — the timing sources interfere with each other and cause call drops. The current setup connects all the PRIs of a single telco to a single dial bank, eliminating that problem. There are currently 3 telcos providing PRIs in the largest centre.

    The client and we are happily running vanilla Asterisk Debian packages with (even though I say so myself) some scripting to die for. Setup is completely stable and is being used to generate some $15M of business annually for the client.


    — Raj