[Asterisk-bsd] current problem on 5.2.1-r-p9

Fri Sep 3 15:52:51 CDT 2004

well I grabbed a clean cvs and built it fine and installs fine but when I run 
it Iget audio from the files but on echo test I get no audio this is the cli> 
line info I get

Sep  3 16:38:55 NOTICE[135398400]: rtp.c:416 ast_rtp_read: RTP: Received 
packet with bad UDP checksum

next is the debug output.
mypbx*CLI> sip debug
SIP Debugging Enabled
mypbx*CLI>

Sip read:

0 headers, 0 lines
mypbx*CLI>

Sip read:
INVITE sip:902 at 10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bK14297bbd3f653a4f
From: <sip:10 at 10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902 at 10.0.0.1>
Contact: <sip:10 at 10.0.0.2>
Call-ID: b63e3b20c8fb2746 at 10.0.0.2
CSeq: 46025 INVITE
User-Agent: Grandstream BT100 1.0.5.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 329

v=0
o=10 8000 8000 IN IP4 10.0.0.2
s=SIP Call
c=IN IP4 10.0.0.2
t=0 0
m=audio 5004 RTP/AVP 98 0 8 18 2 15 4 9
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/8000
a=ptime:40

12 headers, 16 lines
Using latest request as basis request
Sending to 10.0.0.2 : 5060 (non-NAT)
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 15
Found RTP audio format 4
Found RTP audio format 9
Peer audio RTP is at port 10.0.0.2:5004
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G726-32
Found description format G728
Found description format G723
Found description format G722
Capabilities: us - 0x40e(GSM|ULAW|ALAW|ILBC), peer - audio=0x51d(G723|ULAW|
ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0x40c(ULAW|ALAW|ILBC)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x1
(G723)
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
10.0.0.2;branch=z9hG4bK14297bbd3f653a4f;received=10.0.0.2;rport=5060
From: <sip:10 at 10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902 at 10.0.0.1>;tag=as1abe8dc5
Call-ID: b63e3b20c8fb2746 at 10.0.0.2
CSeq: 46025 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902 at 192.168.0.4>
Proxy-Authenticate: Digest realm="asterisk", nonce="544d8e3a"
Content-Length: 0

 to 10.0.0.2:5060
Scheduling destruction of call 'b63e3b20c8fb2746 at 10.0.0.2' in 15000 ms
Found user '10'
mypbx*CLI>

Sip read:
ACK sip:902 at 10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bK14297bbd3f653a4f
From: <sip:10 at 10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902 at 10.0.0.1>;tag=as1abe8dc5
Contact: <sip:10 at 10.0.0.2>
Call-ID: b63e3b20c8fb2746 at 10.0.0.2
CSeq: 46025 ACK
User-Agent: Grandstream BT100 1.0.5.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

11 headers, 0 lines

Sip read:
INVITE sip:902 at 10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bK0b18f0124fbf7ce3
From: <sip:10 at 10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902 at 10.0.0.1>
Contact: <sip:10 at 10.0.0.2>
Proxy-Authorization: DIGEST username="10", realm="asterisk", algorithm=MD5, 
uri="sip:902 at 10.0.0.1", nonce="544d8e3a", 
response="e44ce2c3ac0a11459fd4dcafb6f487cd"
Call-ID: b63e3b20c8fb2746 at 10.0.0.2
CSeq: 46026 INVITE
User-Agent: Grandstream BT100 1.0.5.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 329

v=0
o=10 8000 8000 IN IP4 10.0.0.2
s=SIP Call
c=IN IP4 10.0.0.2
t=0 0
m=audio 5004 RTP/AVP 98 0 8 18 2 15 4 9
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/8000
a=ptime:40

13 headers, 16 lines
Using latest request as basis request
Sending to 10.0.0.2 : 5060 (NAT)
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 15
Found RTP audio format 4
Found RTP audio format 9
Peer audio RTP is at port 10.0.0.2:5004
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G726-32
Found description format G728
Found description format G723
Found description format G722
Capabilities: us - 0x40e(GSM|ULAW|ALAW|ILBC), peer - audio=0x51d(G723|ULAW|
ALAW|G726|G729A|ILBC)/video=0x0(EMPTY), combined - 0x40c(ULAW|ALAW|ILBC)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x1
(G723)
Found user '10'
Looking for 902 in admin
list_route: hop: <sip:10 at 10.0.0.2>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
10.0.0.2;branch=z9hG4bK0b18f0124fbf7ce3;received=10.0.0.2;rport=5060
From: <sip:10 at 10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902 at 10.0.0.1>;tag=as2fb4ad13
Call-ID: b63e3b20c8fb2746 at 10.0.0.2
CSeq: 46026 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902 at 192.168.0.4>
Content-Length: 0

 to 10.0.0.2:5060
We're at 192.168.0.4 port 12454
Answering with preferred capability 0x400(ILBC)
Answering with preferred capability 0x2(GSM)
Answering with preferred capability 0x4(ULAW)
Answering with preferred capability 0x8(ALAW)
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
10.0.0.2;branch=z9hG4bK0b18f0124fbf7ce3;received=10.0.0.2;rport=5060
From: <sip:10 at 10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902 at 10.0.0.1>;tag=as2fb4ad13
Call-ID: b63e3b20c8fb2746 at 10.0.0.2
CSeq: 46026 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902 at 192.168.0.4>
Content-Type: application/sdp
Content-Length: 231

v=0
o=root 46394 46394 IN IP4 192.168.0.4
s=session
c=IN IP4 192.168.0.4
t=0 0
m=audio 12454 RTP/AVP 98 3 0 8
a=rtpmap:98 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

 to 10.0.0.2:5060
mypbx*CLI>

Sip read:
ACK sip:902 at 192.168.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKfd8d676776c9a9a3
From: <sip:10 at 10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902 at 10.0.0.1>;tag=as2fb4ad13
Contact: <sip:10 at 10.0.0.2>
Proxy-Authorization: DIGEST username="10", realm="asterisk", algorithm=MD5, 
uri="sip:902 at 192.168.0.4", nonce="544d8e3a", 
response="92bb701a91a2254ac4845f68c4d94a4c"
Call-ID: b63e3b20c8fb2746 at 10.0.0.2
CSeq: 46026 ACK
User-Agent: Grandstream BT100 1.0.5.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

Retransmitting #4 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.137.52.48:5060;branch=z9hG4bK89364DB16CF4410D98AC9C586A7B28D7;received=217.137.52.48;rport=5060
From: Test <sip:11 at 141.158.116.67>;tag=2235100740
To: <sip:902 at 141.158.116.67>;tag=as7f3a4393
Call-ID: 5D977977-C457-4D78-B1D3-84F994118AFF at 217.137.52.48
CSeq: 15113 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902 at 192.168.0.4>
Content-Type: application/sdp
Content-Length: 231

v=0
o=root 46394 46394 IN IP4 192.168.0.4
s=session
c=IN IP4 192.168.0.4
t=0 0
m=audio 19638 RTP/AVP 98 3 0 8
a=rtpmap:98 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

 to 217.137.52.48:5060
Sep  3 16:47:51 NOTICE[138565632]: rtp.c:416 ast_rtp_read: RTP: Received 
packet with bad UDP checksum
Sep  3 16:47:51 NOTICE[138565632]: rtp.c:416 ast_rtp_read: RTP: Received 
packet with bad UDP checksum
Retransmitting #5 (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.137.52.48:5060;branch=z9hG4bK89364DB16CF4410D98AC9C586A7B28D7;received=217.137.52.48;rport=5060
From: Test <sip:11 at 141.158.116.67>;tag=2235100740
To: <sip:902 at 141.158.116.67>;tag=as7f3a4393
Call-ID: 5D977977-C457-4D78-B1D3-84F994118AFF at 217.137.52.48
CSeq: 15113 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902 at 192.168.0.4>
Content-Type: application/sdp
Content-Length: 231

v=0
o=root 46394 46394 IN IP4 192.168.0.4
s=session
c=IN IP4 192.168.0.4
t=0 0
m=audio 19638 RTP/AVP 98 3 0 8
a=rtpmap:98 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
so at this point outgoing audio from clients conected to * no go but the 
prerecorded play fine... no audio  on echo test