Hello,
Is it possible to do multiple google voice accounts under pjsip ?
Can I use this service with my Obihai 110?I think we're ready for the pioneers now. Complete tutorial available on Nerd Vittles tomorrow. Until then, it is recommended that you only attempt this if you have purchased a HiFormance VPS with Incredible PBX 13-13 support. Other options are the Incredible PBX 13-13 ISO or a CentOS 6 (only!) server running the latest Incredible PBX 13-13. Start there. Once your server is operational (one minute at HiFormance), do the following:
Code: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
Making calls: dial GV prefix (48) plus a 10-digit number
Receiving calls: call your GV number from a smartphone or outside phone
For future reference, your refresh_token is stored in pjsip_custom.conf in /etc/asterisk. You can modify it to assign a new Google Voice number to your account. Then restart Asterisk. Multiple GV numbers are not yet supported, but it's on our radar.
You also can modify the Google Voice behavior for incoming and outgoing calls by editing extensions_custom.conf. At the top of the file, you’ll find the [from-internal-custom]context which controls outbound calling with your Google Voice trunk. If you would prefer to use a different dialing prefix for outgoing Google Voice calls, simply change 48 to the prefix desired in every line of the context. Then reload your Asterisk dialplan by issuing the following command: asterisk -rx "dialplan reload"
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 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"
hi, how do i remove the 48 dial prefix from the extensions_custom.conf file?
i am able to change 48 to any 2 digits, but not remove and not to a single digit?
tks
[from-internal-custom]
;# // BEGIN gvsip outgoing
exten => _NXXNXXXXXX,1,Set(CALLERID(dnid)=${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Set(CALLERID(dnid)=1${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Goto(1${EXTEN})
exten => _1NXXNXXXXXX,1,Set(CHANNEL(accountcode)=Google Voice)
exten => _1NXXNXXXXXX,n,Dial(PJSIP/1${EXTEN:1}@gvsip,,r)
exten => _1NXXNXXXXXX,1,Set(CHANNEL(accountcode)=Google Voice)
exten => _1NXXNXXXXXX,n,Dial(PJSIP/1${CALLERID(dnid):1}@gvsip,,r)
;# // END gvsip outgoing
[from-internal-custom]
;# // BEGIN gvsip outgoing
exten => _NXXNXXXXXX,1,Set(CALLERID(dnid)=${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Set(CALLERID(dnid)=1${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Goto(1${EXTEN},1)
exten => _1NXXNXXXXXX,1,Set(CHANNEL(accountcode)=Google Voice)
exten => _1NXXNXXXXXX,n,Dial(PJSIP/1${EXTEN}@gvsip,,r)
exten => _1NXXNXXXXXX,1,Set(CHANNEL(accountcode)=Google Voice)
exten => _1NXXNXXXXXX,n,Dial(PJSIP/1${CALLERID(dnid)}@gvsip,,r)
;# // END gvsip outgoing
Via: SIP/2.0/TLS 10.32.0.41:5061;rport;branch=RandomStringHere;alias
From: "701" <sip:[email protected]>;tag=RandomStringHere
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: RandomStringHere
CSeq: 22068 INVITE
exten => _1NXXNXXXXXX,1,Set(CHANNEL(accountcode)=Google Voice)
exten => _1NXXNXXXXXX,n,Dial(PJSIP/1${EXTEN}@gvsip,,r)
exten => _1NXXNXXXXXX,1,Set(CHANNEL(accountcode)=Google Voice)
exten => _1NXXNXXXXXX,n,Dial(PJSIP/1${CALLERID(dnid)}@gvsip,,r)
For extension (pattern) _1NXXNXXXXXX you have two steps numbered 1.
It looks like you are trying to place a call to your own callerid ??
You are getting an extra "1" when you dial because you are putting a 1 before the extension in Dial(PJSIP/1${EXTEN}@gvsip,,r)
[from-internal-custom]
;# // BEGIN gvsip outgoing
exten => _NXXNXXXXXX,1,Set(CALLERID(dnid)=${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Set(CALLERID(dnid)=1${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Goto(1${EXTEN},1)
exten => _1NXXNXXXXXX,1,Set(CHANNEL(accountcode)=Google Voice)
exten => _1NXXNXXXXXX,n,Dial(PJSIP/1${EXTEN}@gvsip,,r)
;exten => _NXXNXXXXXX,1,Set(CHANNEL(accountcode)=Google Voice) <---- this line is commented out.
exten => _1NXXNXXXXXX,n,Dial(PJSIP/1${CALLERID(dnid)}@gvsip,,r)
;# // END gvsip outgoing
i find my inbound calls work right after i do a dialplan reload for some strange reason
haven't figured it out why yet
edit, not really the reload, it just sometimes rings but most of the time it doesn't
same => n,Dial(SIP/701,20,D(:1))
You should really just do this as a Custom Trunk (custom dial string = "PJSIP/$OUTNUM$@gvsip") and then set up Outbound Routes in FreePBX to point to that.Thanks - I'm not very well versed in raw asterisk config so I'll give it a shot.
can you share your config (or a sample)? I have a completely vanilla version of everything - default route, one extension... i have uncommented the line per the instructions:
Code:same => n,Dial(SIP/701,20,D(:1))
but still - nothing :\ i dont even see calls hitting the server so I think somehow the obitalk device isnt being routed to by GV
You should really just do this as a Custom Trunk (custom dial string = "PJSIP/$OUTNUM$@gvsip") and then set up Outbound Routes in FreePBX to point to that.
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