Too many sip clients? Do I need Asterisk or Siproxed?

  • Hey because I'm running pfsense now I don't have a way to plug in my old analog phone.
    So I thought why not just install/use a sip client on all the smartphones in the household and use my voip connection that way.

    I tested it on one devices and it work straight out of the gate (not port forwarding required, which seemed strange to me).
    So I entered the sip credentials on the other to smartphones.

    After a few tests it turns out that not all phones were always receiving all calls. I'm not sure why that is, but one theory I've got is that my provider only allows a limited number of devices to connect at once.

    I'm not sure how I would go about solving this Problem.
    Is my theory completely wrong and I just need to adjust some settings, or do I need to use siproxd or asterisk, if so which do I need to use.

    Thanks for the help.

  • You're using the same SIP credentials on them all? That generally won't work, each SIP phone should have (and in cases must have) a unique extension. There is nothing at the firewall level you can do to make something work that won't work with your provider. The various SIP options there are for differing means of handling NAT, which isn't relevant from the sound of things.

  • Yes I'm only using one sip account.
    But there must be a way. Many home routers like the fritzboxs offer a feature like this. You can enable as many sip phones on the box it self and the box then connects to the server.
    So when a call comes in all devices ring.
    This is why I thought the proxy might be the right thing.

    Are there any solutions for this problem within pfsense?
    If not I would always be able to run a VM to solve the Problem.

  • You could accommodate that using Asterisk. Have it connect out to your SIP account, then create individual accounts for every device in it. That's basically what Fritzbox seems to do. Your clients connect only to the local PBX, and only it connects to the provider.

  • Or buy inexpensive Grandstream or Cisco adapter and you can use analog phones with VoIP service. You can get it for less then $50.

  • Ah ok.
    So I can achieve my goal using asterisk, great.
    I'm very new to even the concept of asterisk, but I'll try my luck.
    Could one of you point me in the right direction (a good search term or guide)?

    I thought about using an adapter but because I'm only "playing" around with pfsense to broaden my horizon I don't want to invest too much money in hardware.
    If I wanted an easy solution I would just get an old fritzbox, those are even cheaper :).

  • The only other way would be to get a public IP address for each of your client devices and do 1:1 nat with them, or better yet let each client have its own individual public IP address with no NAT transversal.

  • His problem has nothing with firewall. Problem is in single SIP account and multiple clients. They all must have unique SIP account. I can do that with my provider (Vitelity) as I can create sub-accounts. Currently I have two, one for my desk phone and one for my cell phone. I can configure call routing too and use both device at the same time. I'm still limited to single DID and two calls maximum.

  • If I wanted an easy solution I would just get an old fritzbox, those are even cheaper

    The AVM FB is a combination between router, phone PBX and modem, it splits the services
    by feeding them with different cables from outside you will not have given by the pfSense!

    To connect your old analog phone to the network it could be running over this Cisco adapter!
    Cisco Small Business SPA112 2 Port Phone Adapter

    There are three often used ways to get it flawless working:

    • A STUN Server in the Internet
    • A PBX appliance like Asterisk in the DMZ
    • An integrated SIP-ALG inside of pfSense

    In my eyes the fastest way:
    set up the asterisk pbx appliance in the DMZ
    open the right ports to the pbx appliance
    Connect your phones to the appliance

  • I've not had much luck with Asterisk servers behind pfsense or really any kind of NAT.

  • I've not had much luck with Asterisk servers behind pfsense or really any kind of NAT.

    Put the asterisk PBX appliance inside of a DMZ
    then open some ports, what ever needed to get it running, like an ordinary server with Internet contact
    adjust your rule sets
    thats it

    Ok fairly I must tell you that I am using the Siemens Gigaset DX800a since years
    and this is staying outside of the LAN but there fore I haven´t any kind of problems with VOIP.

  • For me, asterisk server works fine with phones that are also behind pfsense with the server.  The issues always happen when the server is behind pfsense and the phones are elsewhere in the world behind some other router.

  • NAT was not an original feature in SIP. In fact it had to be shoe-horned in later as carriers started going after your run of the mill residential service customer. Its still not perfect and devices and carriers still deal with it in different ways.

    Most carriers will let you sign in with multiple SIP devices into one account. I know of non that can reliably do so when the customer is trying to use the same "public" IP address for more than one device.

    I use VOIPo and have multiple SIP devices on my primary business numbers at different locations.

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