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Vintage Telephony and Telecomms Vintage Telephones, Telephony and Telecomms Equipment |
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31st Mar 2019, 2:52 pm | #21 | |
Pentode
Join Date: Mar 2012
Location: Rotherham, South Yorkshire, UK.
Posts: 152
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Re: Using a slide Switch 2A to switch between POTS and FreePBX
Quote:
What I'm curious about, though, is:- a) when the time condition is NOT matched, there seems to be no way to leave the call alone? i.e. ignore it and let it filter through the normal inbound call route? I have two trunks now working, the BT and the Sipgate. If I get a call on the BT between 6pm and 7pm I'd like to present an IVR to the caller basically saying: "We're putting the kids to bed, press '1' if your call is REALLY urgent and we'll take it" but I can't seem to be able to do that without also telling the sipgate line to basically mirror the behaviour of the BT line which I don't want to do. Does that make sense? It seems like if I tell the time conditions if it's NOT between 6 and 7 then whatever destination I apply for a NON-MATCH will apply to both lines? |
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31st Mar 2019, 3:42 pm | #22 |
Dekatron
Join Date: May 2008
Location: Derby, UK.
Posts: 7,735
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Re: Using a slide Switch 2A to switch between POTS and FreePBX
The BT line and the Sipgate trunk should point towards different contexts within the dialplan, so you can then have different behaviours for each of them.
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If I have seen further than others, it is because I was standing on a pile of failed experiments. |
31st Mar 2019, 4:30 pm | #23 | ||
Dekatron
Join Date: May 2010
Location: Greater Manchester, UK.
Posts: 18,724
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Re: Using a slide Switch 2A to switch between POTS and FreePBX
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So point one of the time-condition destinations to where the inbound route would have sent it if acting alone. Quote:
We lesser mortals using FreePBX have a simple construct called "Inbound Routes to direct the calls on the basis of DID (trunk line) and/or CID (presented number). It sounds to me the reason your system has no discrimination, is that you have accidentally created a route with both of these parameters left blank. It's a common mistake and easily done! If you give me a little time, I will PM you how my landline and Sipgate trunks and inbound routes are set up and I'll leave everything in the clear other than the passwords.
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1st Apr 2019, 5:00 pm | #24 |
Pentode
Join Date: Mar 2012
Location: Rotherham, South Yorkshire, UK.
Posts: 152
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Re: Using a slide Switch 2A to switch between POTS and FreePBX
Thanks, Graham, I've replied under separate cover (PM!) re: the config settings!
What confuses me about the inbound routes is this: I have two inbound routes:- Sipgate BT I have two trunks:- Sipgate BT I have two extensions:- 4001 (this rings my house phones when my BT line rings) 4002 (this rings a softphone when my Sipgate line rings) I don't see how I can use a time condition to ONLY apply to the BT inbound route without also affecting the Sipgate route. I'll do some more playing tonight. As a corollary, has anyone using Freepbx ever come across an issue where calls just simply won't work from certain sources? I can ring my Freepbx instance from a mobile, from work, from Sipgate - all works fine - but today Npower were trying to get hold of us and the call just kept failing. When you pick up the phone you just get bizarre noises like when a Skype call goes slightly pear shaped. No voice could be heard. (When I get home I'll have a look at the logs.) |
1st Apr 2019, 9:39 pm | #25 | ||
Dekatron
Join Date: May 2010
Location: Greater Manchester, UK.
Posts: 18,724
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Re: Using a slide Switch 2A to switch between POTS and FreePBX
Quote:
BT Trunk >> BT Route >> | Time Condition MET >> IVR | | Time Condition NOT MET >> EXT4001 Sipgate trunk >> Sipgate Inbound Route >> Ext 4002 Quote:
Try calling your BT number from your mobile using the 141 prefix. If that's the issue, let me know.
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-- Graham. G3ZVT Last edited by Graham G3ZVT; 1st Apr 2019 at 9:55 pm. |
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3rd Apr 2019, 10:35 am | #26 |
Pentode
Join Date: Mar 2012
Location: Rotherham, South Yorkshire, UK.
Posts: 152
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Re: Using a slide Switch 2A to switch between POTS and FreePBX
Ahhhh.....
Do I understand correctly that you create a time group, then a time condition...but then you've achieved precisely nothing unless you then insert the time condition into your inbound route? I was under the impression that as soon as you create a time group and time condition that it takes effect...but that's not right, is it? And that's the key to how to leave the other (Sipgate) route unmolested - right?? The fact that if I don't insert the time condition into the Sipgate inbound route, the Sipgate route will behave as if it's not there? |
3rd Apr 2019, 11:17 pm | #27 |
Dekatron
Join Date: May 2010
Location: Greater Manchester, UK.
Posts: 18,724
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Re: Using a slide Switch 2A to switch between POTS and FreePBX
Yes and I am sure you can see now how you can chain together modules to do complex tasks. Sometimes it's useful to draw a flowchart so you don't get lost.
We haven't really mentioned outbound calls, are you using Sipgate (I think they are expensive). Do you need any help with dial rules?
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5th Apr 2019, 8:38 pm | #28 |
Pentode
Join Date: Mar 2012
Location: Rotherham, South Yorkshire, UK.
Posts: 152
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Re: Using a slide Switch 2A to switch between POTS and FreePBX
Hallo! Would LOVE to spend a week setting this up to my heart's content but sadly there are day jobs and children and ....
So I'm only now able to return to this! Outbound dialling works fine. If I prefix '9' I dial out via Sipgate, omit the '9' and I dial out via the PSTN. That's what I want for the minute. Confusingly my Sipgate DID is alphanumeric, so Freepbx won't let me enter it into the Did field. But nevertheless it seems to work anyway!! Going to see if I can get some fancy stuff setup tonight. Or half way fancy. I have semi ambitious goals for a domestic landline! |
6th Apr 2019, 9:24 pm | #29 |
Dekatron
Join Date: May 2010
Location: Greater Manchester, UK.
Posts: 18,724
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Re: Using a slide Switch 2A to switch between POTS and FreePBX
My first foray into Asterisk using an old x86 machine involved "Dial 9 for an outside line" because that's how office telephones usually work, but I quickly realised this was unnecessary and confusing for the uninitiated. Outbound rules can determine routing, and here we touch on something you mentioned earlier, the pause while the system determines if you have finished dialling. Unlike the PSTN, we don't have to strictly parse our digit length for example 123 on my system gives me the Alison Smith speaking clock but 12310, 12311, 12320, 12321 [...] etc. turn on and off relays connected to the GPIO pins. You can always dial # to signal that you've finished
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-- Graham. G3ZVT Last edited by Graham G3ZVT; 6th Apr 2019 at 9:29 pm. |