cannot dial out to GV
cannot dial out to GV
I can receive calls through my google voice number that I have forwarded to my gizmo account.
I cannot dialout, however.
I do not see anything in my sipsorcery console. I get a busy signal from my SPA-3102 after i dial the last number.
below is my syslog:
syslog server(port:514) started on Wed Oct 27 00:05:25 2010
[0]Off Hook
2. Report digit 1 (1)(40 ms)
2. Report digit 8 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
2. Report digit 2 (1)(40 ms)
2. Report digit 2 (1)(40 ms)
2. Report digit 5 (1)(40 ms)
2. Report digit 2 (1)(40 ms)
2. Report digit 5 (1)(40 ms)
2. Report digit 2 (1)(40 ms)
2. Report digit 5 (1)(40 ms)
Calling:18002252525@sipsorcery.com:0
[0:0]AUD ALLOC CALL (port=16424)
[0:0]RTP Rx Up
[0]->69.59.142.213:5060(1069)
[0]->69.59.142.213:5060(1069)
INVITE sip:18002252525@sipsorcery.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5070;branch=z9hG4bK-805b6fae;rport
From: Anonymous <sip:anonymous@localhost>;tag=4d3335aa665674d9o0
To: <sip:18002252525@sipsorcery.com>
Remote-Party-ID: <sip:MYUSERNAME@sipsorcery.com>;screen=yes;privacy=full;party=calling
Call-ID: ef338ebe-1d91ced5@localhost
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Anonymous <sip:anonymous@192.168.0.3:5070>
Expires: 240
User-Agent: MagicJack/2.0.562d (SJ Labs)
Content-Length: 436
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 3068 3068 IN IP4 192.168.0.3
s=-
c=IN IP4 192.168.0.3
t=0 0
m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[0]<<69.59.142.213:5060(295)
[0]<<69.59.142.213:5060(295)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3:5070;branch=z9hG4bK-805b6fae;rport=5070;received=my.external.ip
To: <sip:18002252525@sipsorcery.com>
From: "Anonymous" <sip:anonymous@localhost>;tag=4d3335aa665674d9o0
Call-ID: ef338ebe-1d91ced5@localhost
CSeq: 101 INVITE
Content-Length: 0
[0]<<69.59.142.213:5060(381)
[0]<<69.59.142.213:5060(381)
SIP/2.0 404 NotFound
Via: SIP/2.0/UDP 192.168.0.3:5070;branch=z9hG4bK-805b6fae;rport=5070;received=my.external.ip
To: <sip:18002252525@sipsorcery.com>
From: "Anonymous" <sip:anonymous@localhost>;tag=4d3335aa665674d9o0
Call-ID: ef338ebe-1d91ced5@localhost
CSeq: 101 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, SUBSCRIBE
[0]->69.59.142.213:5060(400)
[0]->69.59.142.213:5060(400)
ACK sip:18002252525@sipsorcery.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5070;branch=z9hG4bK-805b6fae;rport
From: Anonymous <sip:anonymous@localhost>;tag=4d3335aa665674d9o0
To: <sip:18002252525@sipsorcery.com>
Call-ID: ef338ebe-1d91ced5@localhost
CSeq: 101 ACK
Max-Forwards: 70
Contact: Anonymous <sip:anonymous@192.168.0.3:5070>
User-Agent: MagicJack/2.0.562d (SJ Labs)
Content-Length: 0
[0]FM Alert Stop RxTx (c=0024e5e8;a=0)
[0:0]AUD Rel Call
CC:Failed w/ Calling
[0]On Hook
[0]->69.59.142.213:5060(4)
[0]->69.59.142.213:5060(4)
0000
I cannot dialout, however.
I do not see anything in my sipsorcery console. I get a busy signal from my SPA-3102 after i dial the last number.
below is my syslog:
syslog server(port:514) started on Wed Oct 27 00:05:25 2010
[0]Off Hook
2. Report digit 1 (1)(40 ms)
2. Report digit 8 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
2. Report digit 0 (1)(40 ms)
2. Report digit 2 (1)(40 ms)
2. Report digit 2 (1)(40 ms)
2. Report digit 5 (1)(40 ms)
2. Report digit 2 (1)(40 ms)
2. Report digit 5 (1)(40 ms)
2. Report digit 2 (1)(40 ms)
2. Report digit 5 (1)(40 ms)
Calling:18002252525@sipsorcery.com:0
[0:0]AUD ALLOC CALL (port=16424)
[0:0]RTP Rx Up
[0]->69.59.142.213:5060(1069)
[0]->69.59.142.213:5060(1069)
INVITE sip:18002252525@sipsorcery.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5070;branch=z9hG4bK-805b6fae;rport
From: Anonymous <sip:anonymous@localhost>;tag=4d3335aa665674d9o0
To: <sip:18002252525@sipsorcery.com>
Remote-Party-ID: <sip:MYUSERNAME@sipsorcery.com>;screen=yes;privacy=full;party=calling
Call-ID: ef338ebe-1d91ced5@localhost
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Anonymous <sip:anonymous@192.168.0.3:5070>
Expires: 240
User-Agent: MagicJack/2.0.562d (SJ Labs)
Content-Length: 436
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 3068 3068 IN IP4 192.168.0.3
s=-
c=IN IP4 192.168.0.3
t=0 0
m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
[0]<<69.59.142.213:5060(295)
[0]<<69.59.142.213:5060(295)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.3:5070;branch=z9hG4bK-805b6fae;rport=5070;received=my.external.ip
To: <sip:18002252525@sipsorcery.com>
From: "Anonymous" <sip:anonymous@localhost>;tag=4d3335aa665674d9o0
Call-ID: ef338ebe-1d91ced5@localhost
CSeq: 101 INVITE
Content-Length: 0
[0]<<69.59.142.213:5060(381)
[0]<<69.59.142.213:5060(381)
SIP/2.0 404 NotFound
Via: SIP/2.0/UDP 192.168.0.3:5070;branch=z9hG4bK-805b6fae;rport=5070;received=my.external.ip
To: <sip:18002252525@sipsorcery.com>
From: "Anonymous" <sip:anonymous@localhost>;tag=4d3335aa665674d9o0
Call-ID: ef338ebe-1d91ced5@localhost
CSeq: 101 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, SUBSCRIBE
[0]->69.59.142.213:5060(400)
[0]->69.59.142.213:5060(400)
ACK sip:18002252525@sipsorcery.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:5070;branch=z9hG4bK-805b6fae;rport
From: Anonymous <sip:anonymous@localhost>;tag=4d3335aa665674d9o0
To: <sip:18002252525@sipsorcery.com>
Call-ID: ef338ebe-1d91ced5@localhost
CSeq: 101 ACK
Max-Forwards: 70
Contact: Anonymous <sip:anonymous@192.168.0.3:5070>
User-Agent: MagicJack/2.0.562d (SJ Labs)
Content-Length: 0
[0]FM Alert Stop RxTx (c=0024e5e8;a=0)
[0:0]AUD Rel Call
CC:Failed w/ Calling
[0]On Hook
[0]->69.59.142.213:5060(4)
[0]->69.59.142.213:5060(4)
0000
Re: cannot dial out to GV
You need to enter your credentials into your ATA/phone. The sipsorcery server isn't going to put your call through if it comes in as anonymous.
Code: Select all
From: Anonymous <sip:anonymous@localhost>;tag=4d3335aa665674d9o0
Re: cannot dial out to GV
I've entered my credentials (under Proxy and Registration) so I don't know why it's sending it as anonymous.
do you know which section I need to change?
thanks.
do you know which section I need to change?
thanks.
Re: cannot dial out to GV
First, make sure you enter it under "Line 1". Second, check if you're sending Line 1 calls thru default gateway (no specific gateway in Line1 dialplan).
Re: cannot dial out to GV
I've got it in line 1, and I don't have anything in the gateway.
my dial plan is:
(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
my dial plan is:
(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
Re: cannot dial out to GV
Subscriber information / Use auth ID -> no
Re: cannot dial out to GV
same thing
do I need to just reset the device?
do I need to just reset the device?
Re: cannot dial out to GV
Probably the easiest way is to reset it to factory defaults and then reconfig for Sipsorcery.
Re: cannot dial out to GV
that worked and I had to fix my dial plan.
I've got it working now. thanks.
I've got it working now. thanks.
Re: cannot dial out to GV
What dial plan worked? I've got the same one in my SPA2102 and I can't make outbound calls. I receive calls OK after I put my WRVS4400N router that's in front to DMZ to this device, but every time I try to call out, I get 1 ring and then a busy tone. I've been banging my head on a wall trying to get this working and can't figure out what the problem is.