"One Line" with SIP Provider; Call Park? Transfer Different Context? Need Concrete Answers

[threehappypenguins]

I am trying to determine how I can free up my SIP trunk so that it can receive additional callers, because I am limited to “one line” with my SIP provider. I want to create a “listen line” for a few people to be able to call in and listen to my church’s live stream. I cannot go to another SIP provider (freephoneline.ca) as I already have this number and I don’t pay a bill for it. I also have a free Callcentric account (1777####### type numbers with the ability to add extensions; so like 1777#######101, 1777#######102, etc).

Is it, or is it not possible to be able to have Asterisk answer a call, put the caller into call park permanently, listening to moh, and then after the caller is in the parking lot, hang up from the SIP trunk, thus freeing it so it can take another call and do the same?

OR, have Asterisk answer a call, automatically transfer/fork to Callcentric, and then hang up from the SIP end, thus freeing it so it can take another call and do the same?

I keep having people tell me that I need a SIP provider with multiple inbound lines, but no explanation as to whether or not I can do what I explained above (without needing multiple lines). However, I have found this post where the person had the same issue as me (“one line”), and was able to fork the call to a smartphone and computer (linphone). I also own an Obi 202 ATA, and in the past I was able to fork incoming calls (same SIP provider, freephoneline) so that it rings my “home phone” with the ATA as well as my smartphone which had a Callcentric extension registered. On the Obi, the main Callcentric # was registered so that it called out on another service port. I also could call in (again, same SIP provider) and have the auto-attendant answer and could use another service provider to call out (such as Google Voice), in order to avoid long distance charges on my cell phone. Asterisk is far, far more customizable than my Obi 202.

I am thinking that this must be possible, and if it is not, can somebody please explain why not?

[jcolp]

As to whether it would work or not depends on the behavior of the underlying provider. If they only allow a single call, then if the call is passing through their platform in any way they may enforce that limit. It’s policy that they’ve dictated, not an inherent limitation or functionality of SIP itself.

For Asterisk if the call is parked in it, then the call has to remain up for it to be there. You can’t hang up from the SIP trunk, because the call is being carried over it. The same applies if it is transferred, even if you push it to the provider. That still consumes an inbound resource on their side.

[david551]

Gven that SIP providers don’t set up direct media paths, most of their processing cost for a call will be in handling the media, so, if you are pushing media through the provider, you can expect them to charge you for it, and to impose throughput limits based on it.