Ok update I decided to try the last stupid thing I should have tried yesterday “reboot -h now” when the system came back up I called into a voicemail box and left a message and low and behold it transcribed it.
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Sendmail-gcloud script not transcribing during run
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Digium D80 vs. Contact Manager?
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Remote Site
Thanks for the reply, i will give that a try. There is no way to have an analog line card at a different site tied directly into the main PBX?
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Remote Site
You can certainly do that. Using an FXO gateway such as
http://www.grandstream.com/products/gateways-and-atas/voip-gateways/product/gxw4104/4108
you wouldn’t even need a server at the remote site (the phones could connect to the headquarters PBX over the VPN).
However, in most applications that would not be a good design. POTS lines are generally much more expensive and less flexible than SIP trunks, but are often chosen because they are more reliable and are less likely to have voice quality issues. Having the remote site connect over the VPN negates those advantages – the POTS lines would be inaccessible in an internet outage and could have quality issues when the VPN is overloaded.
We could give better advice if you provide more details. What country are you in? How many extensions at each site? Number of POTS lines at each site? Are the POTS lines actual copper fed from a central office, or are they delivered over the same fiber or cable with your internet connection?
Do you also have SIP trunks? What kind of phone system does the remote site have now?
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No calls, probably problem with ports opening
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Remote Site
Current design is as such:
Site1 - FreePBX server, 4 POTS lines, 20 extensions.
Site2 - Nortel Analong, 3 POTS lines, 8 handsets.
At each site the copper cxn is delivered with the internet from the provider. We are needing to upgrade site2 in order to provide voicemail and deliver a more robust solution. We are looking to provide dialtone but need to maintain the DIDs that are mapped to the POTS lines.
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EVeryone is busy/congested
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Remote Site
Perhaps a Grandstream GXW4104 as already suggested, available for under $200 on amazon. Similar devices from Patton, Audiocodes,Cisco and Sangoma, they all work much the same way providing a SIP trunk at the FreePBX to the PSTN connections at the Nortel end, if the handsets are analog (plain old telephones) add 8 ports of FXS as SIP extensions on the FreePBX
Generically you are looking for a 4-port FXO (means a trunk not a phone) SIP ATA (means analog telephone adapter) gateway, specifically you will need to get one that is as simple as necessary for you to setup.
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FreePbx to Vocera
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Outbound Caller ID batch update
so should I still do the fwconsole?
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Let's Encrypt - Can't generate new cert, ACME v1 EOL per LE
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Cisco SPA phones, modify EPM to support HTTPS as a provisioning option
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Filestore , S3
Thanks. Do you have any idea about using this feature for other path like recording?
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Inbound calls via same SIP trunk
hi,
Yes you are correct, for each new number you must add a new SIP trunk to the provider.
The dont offer a SIP trunk with as many channels as you want and add numbers above that
Its crazy right ?
What kind of SIP service is that ?
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Sipsettings.class.php error on reload
After the upgrade everything worked okay but today I tried to reload configuration with fwconsole reload and got this output:
fwconsole reload
Reload Started
In Sipsettings.class.php line 433:
json_decode() expects parameter 1 to be string, array given
reload [–json] [–dry-run] [–skip-registry-checks] [–dont-reload-asterisk]
I did not modified anything there and line 433 states:
$pjsip_identifers_order = json_decode($pjsip_identifers_json, true);
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CDR using "start" column instead of "calldate" column in database
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Sipsettings.class.php error on reload
Please make sure you are on latest edge release of framework and sipsettings.
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Freepbx with fritzbox
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FreePBX architecture
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FreePBX with 2N voiceblue
Hi all,
Appreciate if anyone could have a look at this.
I have a freepbx on a remote location with a 2N voiceblue gateway that is used for national calls which has a sim card inserted.
The FreePBX has 2 PJSIP trunk with another provider which is working fine.
the main issue is between the freepbx and the 2N voice blue.
I have a single unauthenticated PJSIP trunk between the freepbx and 2N voicelbue, which is used for national calls. the Freepbx receives the call from the ip phone, if its a national call, it then sends out of this trunk which goes through the 2N voiceblue.
And i have an PJSIP Extension configured on Freepbx to register the 2N voiceblue, this extension is authenticated and is used for the incoming calls on the sim card. when there is an incoming call on the sim card, the 2N voiceblue receives it, and then it should send the call over that Ext to FreePBX which is then sent to the final ip phone.
the problem here is, this PJSIP ext never registers. on the 2N voiceblue page, it shows “registeration error” error code 404. however, the outgoing calls work just fine, except the incoming.
but when i change the Ext driver in freepbx from PJSIP to normal SIP, and change the port from 5060 to 5160 on 2N voiceblue, this ext registers fine, the incoming calls start work, however the outgoing calls Fail !
thank you in advance for any help !
Regards,
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