I am fairly new to SIP telephony and have a question.
My scenario is: i am implementing an IVR system that makes automatic outbound calls.
I have already implemented a prototype with the open source softphone peers.
What i need: I need to make parallel calls with one sip number/account and that was not possible with that softphone client. So i am looking for a way to achieve that requirement.
Is Asterisk the right choice to solve this problem? I would use it as a kind of “middleware” between the softphones and the siptrunk (I think it’s Metaswitch).
That’s a commercial question, not a technical one. You need to ask the organisation with which you have an account as to what limits they impose. Accounts are not SIP constructs. Asterisk imposes no hard limits as to the number of concurrent calls to a domain, or the number of using the same set of authorisation credentials (note that SIP does not require the use of credentials; that is a local policy matter).
Also SIP doesn’t require numbers. There doesn’t have to be a user part of the URI, and the user part does not have to be numeric. I suspect you really mean a PSTN number. As with the PSTN itself, and email, outgoing calls don’t need to have a reply-able URI associated with them, although STIR/SHAKEN may have changed this, when the destination is a PSTN number.
Although the facility is little used, and can be risky, SIP also doesn’t require accounts; the protocol allows a direct IP path to the destination phone, or PABX.
It is SIP, I already implemented a prototype and analyzed the SIP traffic.
The clients should always see the same number calling them. Even if multiple calls in parallel are made.
I wasn’t suggesting it wasn’t SIP; I was contrasting things that are of the nature of SIP, compared with those which are the results of using commercial SIP to PSTN gateways (commonly called Internet Telephony Service Providers).
Where did the clients come from. I assume you mean business clients as client doesn’t make sense in a SIP context, here. I’m not sure why the commercial relationship is relevant here. SIP allows the the same From URI to be sent on unlimited numbers of concurrent calls. Restricting the user part to numbers, ignoring the domain part, and limit numbers of calls are alls limitations imposed by the service provider and not by SIP. (Asterisk makes it easier to present the same domain name on all calls, but that isn’t an absolute restriction.)
(In practice, these days, the problem is with presenting anything other than the number associated with the service account, not with multiple presentation of the same number.)
What is likely to determine whether the provider supports parallel calls is whether their target market is medium to large businesses, as against one or two person businesses (and domestic users), i.e. whether or not they expect to be accessed through a PABX.
The provider supports the same from URI for concurrent calls.
My problem was, my softphones generate different ports in the Contact header for each call
… which leads to the calls don’t work, if the same sip account is used concurrently (with different ports on the same client machine)
so my provider suggested to find a solution that can handle this problem on the “softphone side”.
now i would like to use Asterisk as an automated dialer and IVR system (if that solves my problem).
I presume that the provider is requiring registration, will only accept a single registration, and uses the full address, from the registration, to authenticate outbound calls. If required to register, Asterisk will only register once with the endpoint, and, in any case, will only use the address and port configured for the transport, so those will be fixed, for an endpoint. As such, Asterisk should work if you have correctly stated what the provider allows.
Authenticating outbound calls against the registration, is a fairly onerous restriction.
If the provider is actually checking the Contact header, you may have problems, as using that for authentication is an abuse of the protocol.
If you are getting multiple port numbers from a single client, are you sure that your router isn’t rewriting the SIP. SIP-ALG should be turned off.
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