08
MAR 2011

Posted by rob@junctionnetworks.com at 11:28 AM EST

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Google Voice SIP Address No Longer Available: sip.voice.google.com Now Silent

SIP access to Google Voice Numbers is no longer available. As it was reported and discussed by Todd Vierling, Alok Saboo, and Dan York over the past couple of days, access to Google Voice numbers WAS available in the following format:

SIP:+1{insert GV number}@sip.voice.google.com

This capability is no longer available. The domain is no longer accepting SIP traffic. See below for unanswered INVITES:

U 2011/03/08 10:39:47.365661 66.227.100.25:5060 -> 74.125.155.192:5060
INVITE sip:1914xxxxxxx@sip.voice.google.com SIP/2.0.
Record-Route: sip:66.227.100.25;lr;ftag=A86A8765-B82111BA;nc=1;did=23d.b0320e75;pr=1>.
Via: SIP/2.0/UDP 66.227.100.25;branch=z9hG4bKd48b.fa795d53.0.
Via: SIP/2.0/UDP 10.130.0.105;rport=5060;received=74.92.235.49;branch=z9hG4bK87b01c93E3B90BC0.
From: "Mike Oeth" sip:mike@junctionnetworks.com>;tag=A86A8765-B82111BA.
To: sip:1914xxxxxxx@sip.voice.google.com>.
CSeq: 1 INVITE.
Call-ID: 82cc9d51-40864e96-7583851f@10.130.0.105.
Contact: sip:mike*74.92.235.49!5060_n@66.227.100.25;gr>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.1.0769.
Accept-Language: en.
Supported: 100rel,replaces.
Allow-Events: talk,hold,conference.
Referred-By: sip:call-setup@onsip.com;id=Chz23NcH>.
Max-Forwards: 69.
Content-Type: application/sdp.
Content-Length: 250.
.
v=0.
o=- 1299591439 1299591439 IN IP4 10.130.0.105.
s=Polycom IP Phone.
c=IN IP4 66.227.100.103.
t=0 0.
a=sendrecv.
m=audio 38348 RTP/AVP 9 0 8 127.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:127 telephone-event/8000.

Invite [Expand/Collapse]

Invite [Expand/Collapse]







<1914xxxxxxx>























Comments

Still working for me...

Rob,

I'm still able to make calls to my GV number via the SIP address. I just tried it before writing this comment and it's working perfectly fine for me. But I've heard multiple reports of people either completely unable to get it to work... or who are having intermittent problems.

The net of it is that this is obviously not a "production-ready" service from Google. Let's hope, though, that they do come through with some type of SIP service (and even complete the picture so that we can have our GV number make outbound SIP connections, too).

Dan

Thanks for the reply, Dan

We're taking a poll on Twitter just to see. Still not working for us and others. Thanks again

Still not responsive for us

We received feedback from others that they are no longer able to dial their SIP URI. We have since attempted to dial using different softphones as well sending INVITES.... No response. Looks like it's not working for most anymore

Free GV setup

Alternative SIP methods for Roaming (abroad) Method 1: GV --> IPkall 425 --> LP sip # --> Mobile Fring / Any sip client (X-lite) Method 2: GV --> SipGate US # --> SG sip # --> Mobile Fring / Any sip client (X-lite) Method 3: GV --> LP US DDI # --> LP sip # --> Mobile Fring / Any sip client (X-lite) Ref: http://goo.gl/WasF

PBXes.Org has stable GV routing

I found this through the latest sipdroid release, but it certainly should work for anyone. It's quite easy to set up a PBXES account that will route calls to google voice. You simply set up a trunk in pbxes that routes to Google Talk on the GV end, and then your pbxes sip address routes the call completely as sip to your GV client. Adds an intermediate SIP address, but there's no PSTN involved, and it works well.

Different dan, how do I

Different dan, how do I route calls to Google talk and after I do that does it take.care of the rest? I mean will that step get the calls forwarded to my cell phone?

GV Sip Still working for me

AS of 5pm on 7/7/2011 calling a GV number via sip works just fine for me.

sip calling on HTC G2

Could you post instructions on how you setup your phone to work with gv sip? Thanks.

yate

You know, you can also install a Yate server to be your own sip-gtalk gateway.

You can also use asterisk,

You can also use asterisk, which is what pbxes.org backend is based on.

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