taf2 / telephone

Automatically exported from code.google.com/p/telephone
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Incoming Calls OK Outbound Calls Forbidden #432

Open GoogleCodeExporter opened 8 years ago

GoogleCodeExporter commented 8 years ago
What steps will reproduce the problem?
1. Calling anyone, with full number or two digit extension.

What is the expected output? What do you see instead?
I just get a forbidden feedback.

What version of the product are you using? On what operating system?
1.0.2 on OSX 10.7

Please provide any additional information below.
Here's the log with error..

 Marker - Dec 22, 2011 5:12:33 PM
a: SIP/2.0/UDP 
192.168.9.121:53710;received=74.103.134.18;branch=z9hG4bKPjzgiGTEyImxbzy8JOXlDUm
hGseACKs439;rport=53710
From: "Robert Granholm" 
<sip:4842709045s@altvoip.com>;tag=DFf02X01oEuPLGmntUgvpr.pzvoI6p.o
To: "Robert Granholm" 
<sip:4842709045s@altvoip.com>;tag=aprq45070e1-e9d0mh0008gs0
Call-ID: hguDXJE.-7FZyv.HZwrKJBxUT5NxWY8t
CSeq: 16654 REGISTER
Contact: <sip:4842709045s@192.168.9.121:53710;ob>;expires=45

--end msg--
 17:12:10.850    pjsua_acc.c  SIP outbound status for acc 0 is not active
 17:12:10.850    pjsua_acc.c  Robert Granholm <sip:4842709045s@altvoip.com>: registration success, status=200 (OK), will re-register in 45 seconds
 17:12:10.850    pjsua_acc.c  Keep-alive timer started for acc 0, destination:4.78.147.153:5060, interval:15s
 17:12:36.402  pjsua_media.c  Opening sound device PCM@16000/1/20ms
 17:12:36.409  ec0x106f6f8a0  AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=20 ms
 17:12:36.410   pjsua_call.c  Making call with acc #0 to <sip:28@altvoip.com>
 17:12:36.410  pjsua_media.c  Media index 0 selected for call 5
 17:12:36.412   pjsua_core.c  TX 1117 bytes Request msg INVITE/cseq=5162 (tdta0x10113c600) to UDP 4.79.132.201:5060:
INVITE sip:28@altvoip.com SIP/2.0
Via: SIP/2.0/UDP 
192.168.9.121:53710;rport;branch=z9hG4bKPjc1aPtorMYptDsFQqHUqLD.0H.bdELH8W
Max-Forwards: 70
From: "Robert Granholm" 
<sip:4842709045s@altvoip.com>;tag=L0NgGm.Ldm8AiL-ow3JOAmYoZ8HkyrQZ
To: <sip:28@altvoip.com>
Contact: "Robert Granholm" <sip:4842709045s@192.168.9.121:53710;ob>
Call-ID: DsHS.KRCg7C5ExGxQ4blT-KSmIPJG5kS
CSeq: 5162 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, 
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.2
Content-Type: application/sdp
Content-Length:   462

v=0
o=- 3533580756 3533580756 IN IP4 192.168.9.121
s=pjmedia
c=IN IP4 192.168.9.121
t=0 0
a=X-nat:0
m=audio 4010 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4011 IN IP4 192.168.9.121
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
 17:12:36.412 AKSIPUserAgent  Call 5 state changed to CALLING
 17:12:36.459   pjsua_core.c  RX 324 bytes Response msg 100/INVITE/cseq=5162 (rdata0x100a3fa28) from UDP 4.79.132.201:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
192.168.9.121:53710;received=74.103.134.18;branch=z9hG4bKPjc1aPtorMYptDsFQqHUqLD
.0H.bdELH8W;rport=53710
From: "Robert Granholm" 
<sip:4842709045s@altvoip.com>;tag=L0NgGm.Ldm8AiL-ow3JOAmYoZ8HkyrQZ
To: <sip:28@altvoip.com>
Call-ID: DsHS.KRCg7C5ExGxQ4blT-KSmIPJG5kS
CSeq: 5162 INVITE

--end msg--
 17:12:36.459   pjsua_core.c  RX 355 bytes Response msg 403/INVITE/cseq=5162 (rdata0x1010f6228) from UDP 4.79.132.201:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 
192.168.9.121:53710;received=74.103.134.18;branch=z9hG4bKPjc1aPtorMYptDsFQqHUqLD
.0H.bdELH8W;rport=53710
From: "Robert Granholm" 
<sip:4842709045s@altvoip.com>;tag=L0NgGm.Ldm8AiL-ow3JOAmYoZ8HkyrQZ
To: <sip:28@altvoip.com>;tag=aprqngfrt-2vbtun10005k2
Call-ID: DsHS.KRCg7C5ExGxQ4blT-KSmIPJG5kS
CSeq: 5162 INVITE

--end msg--
 17:12:36.459   pjsua_core.c  TX 370 bytes Request msg ACK/cseq=5162 (tdta0x1010f4c00) to UDP 4.79.132.201:5060:
ACK sip:28@altvoip.com SIP/2.0
Via: SIP/2.0/UDP 
192.168.9.121:53710;rport;branch=z9hG4bKPjc1aPtorMYptDsFQqHUqLD.0H.bdELH8W
Max-Forwards: 70
From: "Robert Granholm" 
<sip:4842709045s@altvoip.com>;tag=L0NgGm.Ldm8AiL-ow3JOAmYoZ8HkyrQZ
To: <sip:28@altvoip.com>;tag=aprqngfrt-2vbtun10005k2
Call-ID: DsHS.KRCg7C5ExGxQ4blT-KSmIPJG5kS
CSeq: 5162 ACK
Content-Length:  0

--end msg--
 17:12:36.459 AKSIPUserAgent  Call 5 is DISCONNECTED [reason = 403 (Forbidden)]
 17:12:36.979  ec0x106f6f8a0  Buffer size adjusted from 2241 to 1762 (eff_cnt=1440)
 17:12:36.993  ec0x106f6f8a0  Buffer size adjusted from 1762 to 128 

Original issue reported on code.google.com by rgranh...@insigniam.com on 22 Dec 2011 at 10:13