ALERT GV: The Sky Has Fallen... Really

wardmundy

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NAF got us going with Incredible PBX on the RasPi platform this afternoon. So I'll write it up for next week. Exciting times!
 

yozh

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Hello,

Is it possible to do multiple google voice accounts under pjsip ?
 

Jebs2k

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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
 

Jeff Dodge

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Not sure if this is best place to report issues.
I did a clean hiformance install -- linked up 2 extensions - but I'm unable to call out or see anything on my google voice account for incoming calls. Below are logs of a call attempt to 8005551212 (by dialing 488005551212)
What's my next step in troubleshooting?

[2018-06-30 14:36:09] VERBOSE[2256][C-00000001] netsock2.c: Using SIP RTP TOS bits 184
[2018-06-30 14:36:09] VERBOSE[2256][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[2018-06-30 14:36:09] VERBOSE[2150][C-00000001] pbx.c: Executing [4818005551212@from-internal:1] Set("SIP/702-00000003", "CHANNEL(accountcode)=Google Voice") in new stack
[2018-06-30 14:36:09] VERBOSE[2150][C-00000001] pbx.c: Executing [4818005551212@from-internal:2] Dial("SIP/702-00000003", "PJSIP/18005551212@gvsip,,r") in new stack
[2018-06-30 14:36:09] VERBOSE[2150][C-00000001] app_dial.c: Called PJSIP/18005551212@gvsip
[2018-06-30 14:36:09] DEBUG[2152] res_pjsip_outbound_registration.c: Found matching outbound registration state
[2018-06-30 14:36:10] ERROR[2234] pjproject: sip_msg Header with no vptr encountered!! Current buffer: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 172.245.218.79:5061;rport;branch=z9hG4bKPj825664b1-c579-409f-9f17-28b9d9047c49;alias
From: "702" <sip:[email protected]>;tag=549d9a74-f357-45e7-8c22-7a25f15b3df5
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: 2148c032-eac8-4d17-85bc-308bbe4c4548
CSeq: 18214 INVITE
[2018-06-30 14:36:10] ERROR[2151] pjproject: sip_msg ..Header with no vptr encountered!! Current buffer: ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 172.245.218.79:5061;rport;branch=z9hG4bKPj825664b1-c579-409f-9f17-28b9d9047c49;alias
From: "702" <sip:[email protected]>;tag=549d9a74-f357-45e7-8c22-7a25f15b3df5
To: <sip:[email protected]>;tag=3ba11c11
Call-ID: 2148c032-eac8-4d17-85bc-308bbe4c4548
CSeq: 18214 ACK
[2018-06-30 14:36:10] VERBOSE[2150][C-00000001] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2018-06-30 14:36:10] VERBOSE[2150][C-00000001] pbx.c: Executing [4818005551212@from-internal:3] Progress("SIP/702-00000003", "") in new stack
[2018-06-30 14:36:10] VERBOSE[2150][C-00000001] pbx.c: Executing [4818005551212@from-internal:4] Wait("SIP/702-00000003", "1") in new stack
[2018-06-30 14:36:11] VERBOSE[2150][C-00000001] pbx.c: Executing [4818005551212@from-internal:5] Playback("SIP/702-00000003", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2018-06-30 14:36:11] VERBOSE[2150][C-00000001] file.c: <SIP/702-00000003> Playing 'silence/1.ulaw' (language 'en')
[2018-06-30 14:36:12] VERBOSE[2150][C-00000001] file.c: <SIP/702-00000003> Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
[2018-06-30 14:36:14] VERBOSE[2150][C-00000001] file.c: <SIP/702-00000003> Playing 'check-number-dial-again.ulaw' (language 'en')
[2018-06-30 14:36:17] VERBOSE[2150][C-00000001] pbx.c: Executing [4818005551212@from-internal:6] Wait("SIP/702-00000003", "1") in new stack
[2018-06-30 14:36:18] VERBOSE[2150][C-00000001] pbx.c: Executing [4818005551212@from-internal:7] Congestion("SIP/702-00000003", "20") in new stack
[2018-06-30 14:36:18] WARNING[2150][C-00000001] channel.c: Prodding channel 'SIP/702-00000003' failed
[2018-06-30 14:36:18] VERBOSE[2150][C-00000001] pbx.c: Spawn extension (from-internal, 4818005551212, 7) exited non-zero on 'SIP/702-00000003'
[2018-06-30 14:36:18] VERBOSE[2150][C-00000001] pbx.c: Executing [h@from-internal:1] Macro("SIP/702-00000003", "hangupcall") in new stack
[2018-06-30 14:36:18] VERBOSE[2150][C-00000001] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/702-00000003", "1?theend") in new stack
[2018-06-30 14:36:18] VERBOSE[2150][C-00000001] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-06-30 14:36:18] VERBOSE[2150][C-00000001] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/702-00000003", "0?Set(CDR(recordingfile)=)") in new stack
[2018-06-30 14:36:18] VERBOSE[2150][C-00000001] pbx.c: Executing [s@macro-hangupcall:4] Hangup("SIP/702-00000003", "") in new stack
[2018-06-30 14:36:18] VERBOSE[2150][C-00000001] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/702-00000003' in macro 'hangupcall'
[2018-06-30 14:36:18] VERBOSE[2150][C-00000001] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/702-00000003'
[2018-06-30 14:36:20] WARNING[2234] pjproject: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 32000
 

Jeff Dodge

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Additional issue I wanted to note - likely not related to naf gvpjsip -- I'm getting 19-20 messages a day from fail2ban @ various times (I've been running 6 days and have 117 messages) Today it was @ 1:03 am -- I have 4 messages [Fail2Ban] Asterisk:started 4 Messages [Fail2Ban] Asterisk:stopped 2 each of [Fail2Ban] SSH: stopped and SSH: started.
It's not telling me the service stopped - just the jail:
Message body example:
Hi,

The jail Asterisk has been stopped.

Regards,

Fail2Ban



Not sure if this caused by logrotate (which has a daily cron entry) or something else -- is anybody else seeing this?
 

Bizzybee

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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"
Can I use this service with my Obihai 110?
 

wardmundy

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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

Try this:
Code:
[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
 

Jebs2k

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got it working with this:

[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
 

mcfuzz

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Can someone please hint on how to add a dial plan for international dialing? My previous dialplan simply had 011 - not sure how to integrate that into the customer extension setup...

Thanks!
 

mcfuzz

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I found an interesting bug? Not sure if it is a bug or not but - if you modify the extensions_custom.conf file to remove the prefix, it does not seem to like that on my end; it doesnt even like it if revert the file back to original... touch and boom!

For example - if mine looks like this:

Code:
[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

And I dial say 17776545321 - the invite looks like this:

Code:
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

It seems that an extra 1 is appended! Now if I revert back and use the file with the 48 prefix - the invite looks good BUT asterisks still sending back a "cannot complete as dialed" error.

Any ideas?
 

mcfuzz

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Another thing - it doesn't seem inbound calls work even when un-commenting the line at the bottom of extensions_custom.conf file :( I don't see GV forwarding the calls to me at all (nothing in the asterisk logs)
 

billsimon

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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)
 

mcfuzz

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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)

Thanks - I'm not very well versed in raw asterisk config so I'll give it a shot.

I guess the more pressing issue is why I am unable to receive inbound calls :(
 

mcfuzz

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Got it to work! This is the dial plan that worked for me when removing 48:

Code:
[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


Now - to figure out how to do international... any suggestions? I tried different variations of 011. in the dial plan but even though the invite looks legit, I get the "cannot complete as dialed..." - also no inbound calls working yet :(
 
Last edited:

Jebs2k

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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
 
Last edited:

mcfuzz

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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

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
 
Last edited:

billsimon

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Thanks - I'm not very well versed in raw asterisk config so I'll give it a shot.
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.
 

Jebs2k

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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

pm'd you
 

mcfuzz

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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.

I got outbound calls to work fine without the prefix... but inbound calls are not even routing to my server :\ heck - not even hitting my firewall as far as I can see. Do we have an IP address range that GV uses for obitalk devices?
 

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