How can PSTN line receive more than one call simultaneously to avoid incoming calls loss. I have 3 PSTN inbound routes. I can hear the waiting tone of the new incoming call while in a running call but why it doesn’t forward to another extension automatically? thank you
You can switch between the call you are on and the incoming call waiting by performing a ‘hook flash’ .
.
You can’t have two active calls on the same analog line but you can pay for ‘call forward busy’ to another number,
I’m confused by references to both PSTN and SIP. As already noted, it is physically impossible to have more than one active call on an analogue line. On the other hand, the rules for this for SIP accounts are commercial decisions for the service provider. SIP imposes no limits on the number of simultaneous calls on a single set of authentication parameters (account), or a single destination number (“DID”).
How is Asterisk connected to the PSTN, in terms of the hardware used, and the channel types?
I’m using analogue line as normally understood, i.e. one with baseband audio suitable for the sort of phone traditionally used as a home phone (although also allowing for earth start lines, which are generally only used for PABXes).
Given this is in the SIP Trunk section, I’m going to guess that PSTN here refers to what would be called a “DID” in the SIP provider industry and FreePBX.
In that case, either the phone is generating the call waiting indication, because you either aren’t using a FreePBX queue or ring group, or you haven’t configured it suitably, or the provider is generating it, because they are basically providing a service for single phones, not PABXes - they are not providing a trunk in the traditional sense, even though it is one in FreePBX terms.
If you are actually getting a SIP service, getting a log, as described in:
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug
will probably help us understand how you are configured.
PSTN = Public Switched Telephone Network. It is how carriers switch calls between each other. The PSTN is not a single technology as it is TDM, wireless, IP, SIP, etc.
As well, DID = Direct Inward Dial and has been term since the TDM days before SIP and FreePBX.
And since this is FreePBX/Asterisk, how is it possible to hear a call waiting indicator on an incoming call that has hit the PBX and not connected to a user? Who is hear the call waiting beep?
In the wider in industry, it is used differently from the way it is used in the SIP world. You get the strange concept of having DID with only one valid number, whereas, in the traditional usage, it is only used for incoming lines that can support and forward multiple PSTN numbers. In the SIP world, DID refers to a single number, whereas, outside of it, it refers the concept of forwarding part of the number.
Also, yes I know what PSTN stands for. The issue here is trying to work out what the OP meant by “PSTN Line”.
Weird, 30 years in the wider industry and that has never been the case for me. But sure.
hehe
To the OP, is your PTSN (sic) really a POTS line?
(then we can maybe ‘disagree’ further 
)
My guess is you are trying to avoid losing the (second) incoming call.
Here is what i would do as a work around
- set up a new VOIP number and Trunk
- have your PTSN provider do a call forward to the VOIP number
now basically ass your calls come in the the new VOIP trunk, which can handle multiple calls at the same time
exactly. I asked the provider about this service and I can pay for this . thank you
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.