Forwarded Google Voice Number Will Not Answer

I have a Google Voice Number that I was to forward to my FreePBX. I have linked the number in Google Voice and it forward the call. However in bypassed the Inbound route and somehow goes directly to Ring Group 200.

Normally My calls come to an IVR that plays a please hold message and then will Ring Ring Group 200. I will play my music on hold while it rings.

However When Google call forward to it, I never see the IVR in the call logs. I see 4 CDR logs for each call and the Application it Dial and it goes to 200. I then rings hunt group 200 3 times with no music on hold and then Google takes a message

CDR Records - The one on the bottom is direct dial to SIP trunk and it worked. The top 4 are from 1 call forwarded call by google.

Can you provide call logs? https://sangomakb.atlassian.net/wiki/spaces/SS/pages/31162494/Providing+Great+Debug#Asterisk-Logs---Part-II

You will find that Google Voice will not work when forwarded to an IVR. It will work if you directly answer with a telephone. This is some AI smarts they use to insure you can’t have an IVR answer (as in a business using the number.) This ability is only available on paid Google Workspace accounts.

You can test this by forwarding the GV number directly to a phone and see that it works.

See: My google voice number will not forward into my Zoom IVR (but does forward to my non IVR Zoom line) - Google Voice Community

I don’t know whether this still works, but try setting Pause Before Answer in the Inbound Route to 4 seconds.

If not, GV has evidently gotten “smarter”. If it does work, experiment to find the shortest delay that works reliably. If the delay is still an issue for non GV calls, use pjsip logger to see whether a Diversion header or similar could be used to distinguish GV calls.

I tried a 5 Second Delay and a 10 Second delay, that was a good idea and I can hear the delay between the 1st and 2nd ring, but no luck. I have setup a port request to just bring the number over to my SIP trunk, and I will leave Google Voice behind.

I had an OBITalk for years, for my home phones, and they have done away with it. You haven’t been able to setup new ones for a while and they said it was going to go down in November, but I finally quit working last week. So I am going to move the home number over to my home FreePBX server, and setup an ATA for the home phones. I guess it is nice that I will have e911 now!

Sorry, I missed that you also have to set Signal RINGING to Yes, as well as Pause Before Answer. If that still doesn’t work, paste the Asterisk log for a failed call at pastebin.com and post the link here.

Signal ringing will not work either. I don’t know how Google Voice detects an IVR, even before or after it answers, but it won’t work. Google Voice (free version) just doesn’t have a place with a PBX anymore. When you can get a DID for $0.15/month from places like BulkVS, the hassle with Google just isn’t worth the time.

Here is a trace from the SIP provider:

proto:UDP 2025-01-16T13:49:06.832392-05:00  76.8.29.198:5060 ---> 144.xx.xx.xx:5062

INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 76.8.29.198:5060;branch=z9hG4bKfs88mt20d01g093d97n0.1
From: "MY NAME" <sip:[email protected]:5060>;tag=gK00111719
To: <sip:[email protected]>
P-Asserted-Identity: "My NAME" <sip:[email protected]:5060>
P-Charge-Info: tel:1828482yyyy;Rate=0.0003;Trk=SRV_Main
Call-ID: [email protected]
CSeq: 173722 INVITE
Max-Forwards: 67
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Accept: application/sdp
Contact: <sip:76.8.29.198:5060;did=017.15c65815;transport=udp>
Diversion: <sip:[email protected]:5060>;privacy=off;screen=no;reason=unknown;counter=1
Supported: replaces,timer
Content-Length: 255
Content-Disposition: session; handling=required
Content-Type: application/sdp
Session-Expires: 3000; refresher=uac
Min-SE: 90

v=0
o=- 2436789930 854583 IN IP4 67.231.1.79
s=-
c=IN IP4 67.231.1.79
t=0 0
m=audio 41068 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

proto:UDP 2025-01-16T13:49:06.852928-05:00  144.xx.xx.xx:5062 ---> 76.8.29.198:5060

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 76.8.29.198:5060;rport=5060;received=76.8.29.198;branch=z9hG4bKfs88mt20d01g093d97n0.1
Call-ID: [email protected]
From: "MY NAME" <sip:[email protected]>;tag=gK00111719
To: <sip:[email protected]>
CSeq: 173722 INVITE
Server: IncrediblePBX-17.0.19.23(22.1.0)
Content-Length:  0


proto:UDP 2025-01-16T13:49:06.890153-05:00  144.xx.xx.xx:5062 ---> 76.8.29.198:5060

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 76.8.29.198:5060;rport=5060;received=76.8.29.198;branch=z9hG4bKfs88mt20d01g093d97n0.1
Call-ID: [email protected]
From: "MY NAME" <sip:[email protected]>;tag=gK00111719
To: <sip:[email protected]>;tag=58672583-a88e-420c-b238-741c9aa2c9ca
CSeq: 173722 INVITE
Server: IncrediblePBX-17.0.19.23(22.1.0)
Contact: <sip:144.xx.xx.xx:5062>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, INFO, MESSAGE
Content-Length:  0


proto:UDP 2025-01-16T13:49:07.067493-05:00  144.xx.xx.xx:5062 ---> 76.8.29.198:5060

SIP/2.0 200 OK
Via: SIP/2.0/UDP 76.8.29.198:5060;rport=5060;received=76.8.29.198;branch=z9hG4bKfs88mt20d01g093d97n0.1
Call-ID: [email protected]
From: "MY NAME" <sip:[email protected]>;tag=gK00111719
To: <sip:[email protected]>;tag=58672583-a88e-420c-b238-741c9aa2c9ca
CSeq: 173722 INVITE
Server: IncrediblePBX-17.0.19.23(22.1.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, INFO, MESSAGE
Contact: <sip:144.xx.xx.xx:5062>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 3000;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   282

v=0
o=- 2436789930 854585 IN IP4 144.xx.xx.xx
s=Asterisk
c=IN IP4 144.xx.xx.xx
t=0 0
m=audio 16654 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

proto:UDP 2025-01-16T13:49:07.15154-05:00  76.8.29.198:5060 ---> 144.xx.xx.xx:5062

ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 76.8.29.198:5060;branch=z9hG4bKochb7e302o8hc9j2v7m1.1
From: "MY NAME" <sip:[email protected]>;tag=gK00111719
To: <sip:[email protected]>;tag=58672583-a88e-420c-b238-741c9aa2c9ca
Call-ID: [email protected]
CSeq: 173722 ACK
Max-Forwards: 67
Content-Length: 0


proto:UDP 2025-01-16T13:49:33.639736-05:00  76.8.29.198:5060 ---> 144.xx.xx.xx:5062

BYE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 76.8.29.198:5060;branch=z9hG4bKfs88mt20d01g093d97n0cd0g0cqd2.1
From: "MY NAME" <sip:[email protected]>;tag=gK00111719
To: <sip:[email protected]>;tag=58672583-a88e-420c-b238-741c9aa2c9ca
Call-ID: [email protected]
CSeq: 173723 BYE
Max-Forwards: 67
Content-Length: 0


proto:UDP 2025-01-16T13:49:33.659937-05:00  144.xx.xx.xx:5062 ---> 76.8.29.198:5060

SIP/2.0 200 OK
Via: SIP/2.0/UDP 76.8.29.198:5060;rport=5060;received=76.8.29.198;branch=z9hG4bKfs88mt20d01g093d97n0cd0g0cqd2.1
Call-ID: [email protected]
From: "MY NAME" <sip:[email protected]>;tag=gK00111719
To: <sip:[email protected]>;tag=58672583-a88e-420c-b238-741c9aa2c9ca
CSeq: 173723 BYE
Server: IncrediblePBX-17.0.19.23(22.1.0)
Content-Length:  0



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