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

Sat Sep 4 03:49:38 CDT 2004

There are some bugs, we found before my 2-weeks rest.
I'll post diffs on digium, I hope it'll help. ;)
BR,
    Oryx.
----- Original Message -----
From: "Richard Neese" <bsdtech at runbox.com>
To: <asterisk-bsd at lists.digium.com>
Sent: Friday, September 03, 2004 11:52 PM
Subject: [Asterisk-bsd] current problem on 5.2.1-r-p9


> 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
>
> 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"
>
>
>  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
>
>
> 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
>
> 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>
>
>
>  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
>
> 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
>
>
>
>
> Retransmitting #4 (NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
>
217.137.52.48:5060;branch=z9hG4bK89364DB16CF4410D98AC9C586A7B28D7;received=2
17.137.52.48;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
>
> 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=2
17.137.52.48;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
>
> 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
>
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