Trying to figure out answers to questions that have been buzzing in my head for some time...

[Halea]

For the purpose of these questions let's assume that we have a DID with a provider, which only accepts userid+password registration type interaction, and we have a FreePBX type PBX connected to it, and it is fully operational with call in/out on that DID.

I'm trying to figure out the following:

1) If I connect a softphone to the provider's server on the same DID, using the same method with the same userid and password am I going to create a conflict? (Like the new registration knocking out the earlier one) (Just to be clear the softphone is not connected to the PBX but to the provider's sip server)

2) Assuming that the PBX and softphone can simultaneously register without kicking each other out, are they both going to ring when there is an incoming call on that DID?

3) Again, assuming that the PBX and softphone nicely played with each other, what would happen when one of them properly unregisters itself. Is that going to impact the other one? (Like forcing it to disconnect, etc.)

4) Would any of the answers be different if the second sip client is not a softphone but another PBX?

 

[kenn10]

1) There is a high probability that if you register a soft phone on the same credentials as the PBX, the last registration will be where the DID routes. You would have a conflict. Typically, you can have multiple outbound calls using the same credentials but if registration is required for an inbound call, the last one to register will take over. Remember that registrations renew periodically, so you would have the softphone client and the PBX having control and the DID switching randomly between the devices.

2) Highly unlikely that both will work on inbound.

3) & 4) Altogether unlikely this scenario has any hope of working.

As an alternative, it makes more sense to have the DID go into the PBX. Have the softphone running as an extension on the PBX and create a ring group. Have the ring group ring the extensions in the PBX and the softphone extensions.

 

[tbrummell]

1. Depends of the provider is Chan_SIP, or PJSIP, or something else (FreeSwitch anyone?). PJSIP apparently supports multiple registrations to the same userid, Chan_SIP does not. Freeswitch/sofia, I have no idea. That is a provider question, and as there are many providers, there are many different SIP servers they could be using.
2. Again, depends on 1. I would say that if it's PJSIP and *if* it works properly, it *should* ring both.
3. It should *not* impact the other, if PJSIP. If CHan_SIP, it would depend on which client refreshed last, and how long before the next re-register.
4. No.

 

[tbrummell]

To further on @kenn10 's footnote, use a provider like Voip.ms and use a sub-account for each "device". Send the DID to a Voip.ms Ring Group that rings both devices. Problem goes away.

 

[Halea]

1) There is a high probability that if you register a soft phone on the same credentials as the PBX, the last registration will be where the DID routes. You would have a conflict. ...

2) Highly unlikely that both will work on inbound.

Thank you @kenn10. My current experimentation results are in line with your input. My "SIP protocol reference book" reading material has a different take on it though. I guess this is a theory vs practical implementation thing.
However among the providers that I used in the past, I did come across one or two that accepted multiple registration and simultaneous incoming call ringing, very much like the SIP reference book says it should. I have no idea what back end server they used. One of them might have been built on sipxecs from way back.

 

[Halea]

1. Depends of the provider is Chan_SIP, or PJSIP, ...

Chan_SIP and PJSIP are FreePBX Asterisk specific SIP channel drivers, right? If so, provider back-ends probably don't use either, as I doubt they use asterisk as a back-end switch. But that's good to know.

 

[tbrummell]

Yes they are, and yes they do. Voip.ms for sure, others I'm sure do as well. Callwithus uses Freeswitch, and, I guess that's all I've used.

 

[Halea]

To further on @kenn10 's footnote, use a provider like Voip.ms and use a sub-account for each "device". Send the DID to a Voip.ms Ring Group that rings both devices. Problem goes away.

voip.ms much like callcentric have layer over layer improvements and/or enhancements that differentiate them in the marketplace.
While those niceties justify their "value-added" pricing, they also make them "too unique" or "one-of-a-kind" from a standards-compliance perspective.
I didn't mention in my OP, and probably I should have, but my focus is how things work, or should work, while remaining as close to the standards as possible. And, I certainly don't want to insinuate that those fine providers are not in compliance with standards set forth by the organizational bodies. It's rather that, the current standards still have lots of "gray areas" or "holes" in them, and many providers make their own solution as they need to fill in those gaps.

 

[Halea]

Yes they are, and yes they do. Voip.ms for sure, others I'm sure do as well. Callwithus uses Freeswitch, and, I guess that's all I've used.

You mean voip.ms is built around asterisk? Wow I am impressed! I didn't know that there was any very large scale asterisk deployments. Coming from a different world of switches ... my bad

 

[kenn10]

Not all SIP standards are implemented by various vendors. There are a number of VOIP providers who use Asterisk or FreeSwitch for the back end switch. The PJSIP driver has more features than the Chan_SIP driver in the Asterisk world. Don't expect a perfect "standards based" world when dealing with VOIP providers or IP PBX's. SIP protocol has many aspects and specifications that are rarely fully implemented by anyone.

When it comes to Asterisk based PBX's, take advice from those in these forums. It helps reduce the amount of time spent banging your head against the wall.

 

[tbrummell]

You mean voip.ms is built around asterisk?

Yes, their infrastructure is built on Asterisk.

 

[dhoppy]

1) If I connect a softphone to the provider's server on the same DID, using the same method with the same userid and password am I going to create a conflict? (Like the new registration knocking out the earlier one) (Just to be clear the softphone is not connected to the PBX but to the provider's sip server)

With Vitelity you could do that. Vitelity lets you create two destinations, so you create two sub-accounts and then route the DID to the two sub-accounts (one to the PBX, one to the softphone).

 

[ostridge]

Using on a provider accepting multiple registration 'sipgate.co.uk' and an ATA or an Android, on a high numbered (unique) sip port say 49xxx
1) both PBX and phones register ok and
2) ring ok but PBX on pi3 is slower to ring, possibly because of blacklist and Lenny. inbound answer on any. outbound from any can ring own number - answer on other. If Lenny answers depends on timing whether neither rings, or a phone pings.
3) No
4) Don't think that makes difference.

In your particular case just try it and see how it works.

 

[Halea]

... just try it and see how it works.

Well, I actually did, right before my post, with several providers. Multiple registration with the same credentials caused the last device registering to grab the line starting a ping-pong game between the devices.
But some time ago I had a provider with whom I had a similar experience to yours; I could simply connect several devices with the same credentials and get them to ring simultaneously.
I was trying (and I still am) to figure out if that behavior was "by the book" or not.
Thanks for your input.