Yes, you get one or the other.
Roger that. Thanks.
Yes, you get one or the other.
exten => _1NXXNXXXXXX,n,Dial(PJSIP/1${EXTEN:1}@gvsip,,)
Could you please provide an example of what "remove the r from the dial command" means? Is that an edit to extensions_custom.conf, or something in the FreePBX interface?
I tried removing the ',,r' from the Dial command in extensions_custom.conf, and while that prevented the call from being taken down after 20 seconds, I no longer hear the destination phone ringing from the source phone; just silence until the voicemail message from the destination phone is heard.
Thanks.
On the Regional tab, check the section "Vertical Service Activation Codes" to see if *48 is a reserved code for use by the 2012, Actually, you can probably just delete ALL Vertical Service Activation Codes, since the PBX will be handling everything for you.
Second, be sure your 2012 Dial Plan(s) have a match for handling *xx numbers.
You forgot the warning about not registering your Google Voice DID in more than one place.
No I didn't but wasn't sure how it applied. I have the GV app installed on my phone but that's only active when I send a text or make calls over WiFi.
Anyway for some strange reason it's now all working. Maybe I needed to restart PBX after making the last of my config changes but I did note for inbound 701 was already set as the extension destination. I can make and receive call using my GV number so all is good at the moment. Just got off a 30 min call with a friend and he didn't complain about call quality at all.
Thanks
I followed the tutorial diligently, but I am still not able to make or receive calls. Here is the error I am seeing in the log file:
[2018-07-08 09:45:05] ERROR[4458] chan_pjsip.c: Unable to create PJSIP channel - endpoint 'gvsip' was not found
[2018-07-08 09:45:05] WARNING[4457][C-00000004] app_dial.c: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
The "Asterisk Info" report also shows no PJSip Endpoints:
Asterisk System uptime: 28 minutes, 40 seconds
Last reload: 28 minutes, 40 seconds
Active SIP Channel(s): 2 Active PJSIP Channel(s): 0 Active IAX2 Channel(s): 0
Sip Registry: 1 PJSip Registrations: 0 IAX2 Registry: 1
Sip Peers:
Online: 1
Online-Unmonitored: 0
Offline: 0
Offline-Unmonitored: 0
PJSip Endpoints:
Available: 0
Unavailable: 0
Unknown: 0
IAX2 Peers:
Online: 0
Offline: 0
Unmonitored: 0
I did a fresh IncrediblePBX13 install and then did the GVSIP install as follows:
cd /root
wget http://incrediblepbx.com/gvsip-naf.tar.gz
tar zxvf gvsip-naf.tar.gz
rm -f gvsip-naf.tar.gz
cd gvsip-naf
./install-gvsip.sh
I input my refresh token, username, and phone number when requested. And I have made sure I am logged out of the GV account being used. What else can I try to get this working? Thanks in advance for any suggestions.
You're using the WRONG INSTALLER for the RasPi platform. Go here.
GVSIP is not designed for use with the same Google Voice account connected to your smartphone. Whether you're currently making a call or sending a text message, the GV account is still registered with Google.
You're using the WRONG INSTALLER for the RasPi platform. Go here.
And, start with a fresh install of IncrediblePBX. I learned the hard way that once you install the CentOS GVSIP package, you cannot install the Raspbian GVSIP package on top of it.
If it's an OBi100 series device, it's no longer supported by Obihai.
I built a fresh RasPi the other day, and all of these files and directories were already in place. Were you using the latest build and where did you get it??
ls-al incrediblepbx13.13-raspbian8.zip
-rw-r--r-- 1 root root 1384643640 Jun 2 21:13 incrediblepbx13.13-raspbian8.zip
md5sum ./incrediblepbx13.13-raspbian8.zip
d57e6c7a4a73214cf0982f7a306456ac ./incrediblepbx13.13-raspbian8.zip
Have you read the tutorial?? It explains how to redirect incoming calls to anywhere you like:
To modify the Google Voice behavior for incoming calls, jump to the bottom of extensions_custom.conf. There you’ll find the [from-external-custom] context which controls the routing and processing of incoming calls to your Google Voice trunk. Several examples are provided. By default, the inbound calls are routed to the Demo IVR (3366). If your PBX has a Ring Group 777 and you’d prefer to send the calls there, simply change 3366 to 777. If you would prefer to send the calls to an extension, then comment out the Demo IVR line with a semicolon and uncomment the SIP/701 line while also replacing 701 with the extension desired. If you’d prefer to send incoming calls to a specific Asterisk application, an example is provided to route the calls to the NV Weather ZIP application. Then reload your Asterisk dialplan by issuing the following command: asterisk -rx "dialplan reload"
Link up your team and customers Phone System Live Chat Video Conferencing
Hosted or Self-managed. Up to 10 users free forever. No credit card. Try risk free.
Check your inbox!
We’ve sent you an email. Click on the button in the email body to verify your email address – (if you can not find it, check your spam folder).
Upon verification you will be directed to the 3CX setup wizard.