Ring to Mobile Abandons Deskphone

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CAZFIRE

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Greetings,

I am working on a Ring All group that we can use to contact on-call staff. The setup is fairly straight forward, making use of some "dummy" extensions and the "Ring my mobile simultaneously" option.

The problem I am having is that for whatever reason, the "Ring my mobile simultaneously" doesn't work as intended. It is more of a handoff. The deskphone will ring a couple of times, but will stop ringing as soon as the call is completed to the cell phone.

From what I can gather, 3CX sees the fact that the cell phone started ringing as an answer, and abandons all other ringing extensions. The same thing happens in the ring group. As soon as one cell phone starts ringing, no other extensions are called.

I am 99% sure that our "Ring my mobile simultaneously" worked at some point, but currently as soon as the mobile rings, the deskphone stops.

I am guessing this is a trunk issue but am really not sure where to look, and after searching the community for a couple of hours, I decided to post.

If you have any insight, that would be great.

Thanks!
 
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Hi,

Best to start from the basics.

Just have an Extension-A with a deskphone and Ring my mobile simultaneously setup and ready to go.

In the mobile, add an actual extension instead of a mobile number, say Extension-B.

Then call that Extension-A from another Extension-C and note the behavior.

You would expect A and B to ring at the same time.

Now if you replace the mobile with an external number (i.e your mobile phone) and as soon as the mobile rings the Ext-A stops, then indeed it may appear that your provider reports the call as answered (while they connect you to the actual mobile).

You can also confirm this via capture, if the provider answers back to your PBX with a 200 OK and starts sending autio (even if you are still hearing their progress tone)
 
The deskphone will ring a couple of times, but will stop ringing as soon as the call is completed to the cell phone.

From what I can gather, 3CX sees the fact that the cell phone started ringing as an answer, and abandons all other ringing extensions. The same thing happens in the ring group. As soon as one cell phone starts ringing, no other extensions are called.

What type of trunks are you sending the call out over?

It sounds as if there is answer supervision happening as soon as the call is established to the mobile number, which can happen when using a PSTN gateway. If that is the case, then , yes, the desk phone would stop ringing. As suggested, check the Activity Log to see when the call was considered answered.
 
Sorry just now getting back to this. It has been a loooong week.

Hi,

Best to start from the basics.

Just have an Extension-A with a deskphone and Ring my mobile simultaneously setup and ready to go.

In the mobile, add an actual extension instead of a mobile number, say Extension-B.

Then call that Extension-A from another Extension-C and note the behavior.

You would expect A and B to ring at the same time.

Now if you replace the mobile with an external number (i.e your mobile phone) and as soon as the mobile rings the Ext-A stops, then indeed it may appear that your provider reports the call as answered (while they connect you to the actual mobile).

You can also confirm this via capture, if the provider answers back to your PBX with a 200 OK and starts sending autio (even if you are still hearing their progress tone)

@JohnS_3CX - Yes, I am able to ring two extensions simultaneously, if the mobile number is set as an extension rather than the actual 10 digit mobile number.

@leejor - Yes, our trunk is a T1 that gets channelized through a gateway appliance.

Based on the activity logs, I am seeing a a 200 OK as soon as my mobile rings (I have sanitized this)
Code:
IP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.200.30:5060;branch=z9hG4bK-524287-1---e8a3c803467d6e40;rport=5060;received=10.10.200.30
From: "9285555555"<sip:[email protected]:5060>;tag=93199b4a
To: <sip:[email protected]:5060>;tag=3676453376
Call-ID: HUv-_QfMmI1zPvOhkPphqw..
CSeq: 1 INVITE
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, INFO, UPDATE, REFER, REGISTER
Contact: <sip:[email protected]:5060;transport=udp>
Server: Patton SN4970 1E24V 00A0BA0BB507 R6.10 2018-01-09 H323 RBS SIP M5T SIP Stack/4.2.14.18
Supported: replaces
Content-Type: application/sdp
Content-Length: 267

Is there anything I can do about this? I thought for certain this was working at some point in the past, but I haven't checked it since we went from v15 to v16 a few months ago. (Currently on 16.0.5.611)

Is there something in the trunk configuration that we can change? If not, I am just going to have to deal with it.

Thanks for your input.
 
I would place calls to some other numbers, landlines, and other mobile providers, then check the log to see if you get a 200 OK , at the same point, when the call starts ringing. If you do, it may be an option in your T1 gateway that requires changing, or, an issue with your provider. If you see it only on calls to one mobile company, then they may have an issue.

If you list the model of your gateway, others, familiar with the model, may have some suggestions.
 
Sorry again for the delay.

We are using a Patton SN4970. I didn't post sooner because I didn't actually have access to it, but was able to get into it today. (The vendor that did the initial deployment got me the password)

I do have a config backed up at this point, but haven't spent a lot of time poking around in it yet. I would be glad to hear of any advice the community might have. (I will also be doing some searching in the forum for people using the same gateway)

Thanks
 
Okay, so I have learned bit about Patton and their relationship with 3CX.

It seems that 3CX officially started supporting their "Trinity" OS in v. 16.0.3.525. What I can't seem to find is anyone who has upgraded their Patton device from SmartWare to Trinity within 3CX, and if they had to do anything additional to get it functional, i.e. re-add the gateway so it picks up the new templates.

I could be wrong, but based on the feature set for Trinity, it seems like I might have a better shot at getting this to work how I want utilizing that. Otherwise, I might look at using some SIP trunks, which I would much rather move to over time anyway.

Any thoughts?
 
Hi

If the gateway replies 200 OK the PBX considers the call answered and the internal extension should stop ringing because the call was answered elsewhere.

Trunks are not bad if you can switch over, this also frees up the instance from a physical line and can potentially be deployed in the cloud if you wish to do so (as opposed to gateways that need a local PBX).
We have a list of inter-op tested and supported providers with pre-configured templates in case you are looking to switch: https://www.3cx.com/partners/voip-providers/
 
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