Posted: Fri Sep 03, 2004 12:53 pm Post 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>
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@10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902@10.0.0.1>;tag=as1abe8dc5
Call-ID: b63e3b20c8fb2746@10.0.0.2
CSeq: 46025 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902@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@10.0.0.2' in 15000 ms
Found user '10'
mypbx*CLI>
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@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@10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902@10.0.0.1>;tag=as2fb4ad13
Call-ID: b63e3b20c8fb2746@10.0.0.2
CSeq: 46026 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902@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@10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902@10.0.0.1>;tag=as2fb4ad13
Call-ID: b63e3b20c8fb2746@10.0.0.2
CSeq: 46026 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902@192.168.0.4>
Content-Type: application/sdp
Content-Length: 231
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@141.158.116.67>;tag=2235100740
To: <sip:902@141.158.116.67>;tag=as7f3a4393
Call-ID: 5D977977-C457-4D78-B1D3-84F994118AFF@217.137.52.48
CSeq: 15113 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902@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
Posted: Sat Sep 04, 2004 12:46 am Post subject: [Asterisk-bsd] current problem on 5.2.1-r-p9
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@runbox.com>
To: <asterisk-bsd@lists.digium.com>
Sent: Friday, September 03, 2004 11:52 PM
Subject: [Asterisk-bsd] current problem on 5.2.1-r-p9
Quote:
well I grabbed a clean cvs and built it fine and installs fine but when I
run
Quote:
it Iget audio from the files but on echo test I get no audio this is the
cli>
Quote:
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>
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|
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|
Quote:
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@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@10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902@10.0.0.1>;tag=as2fb4ad13
Call-ID: b63e3b20c8fb2746@10.0.0.2
CSeq: 46026 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902@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@10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902@10.0.0.1>;tag=as2fb4ad13
Call-ID: b63e3b20c8fb2746@10.0.0.2
CSeq: 46026 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902@192.168.0.4>
Content-Type: application/sdp
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
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
Posted: Mon Sep 06, 2004 7:31 am Post subject: [Asterisk-bsd] current problem on 5.2.1-r-p9
I am running 5.3 beta and am not having any audio issues with the latest
cvs.
Chris
On Fri, 2004-09-03 at 21:52, Richard Neese wrote:
Quote:
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>
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@10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902@10.0.0.1>;tag=as1abe8dc5
Call-ID: b63e3b20c8fb2746@10.0.0.2
CSeq: 46025 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902@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@10.0.0.2' in 15000 ms
Found user '10'
mypbx*CLI>
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@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@10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902@10.0.0.1>;tag=as2fb4ad13
Call-ID: b63e3b20c8fb2746@10.0.0.2
CSeq: 46026 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902@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@10.0.0.1>;tag=aebff69a2f1aebe4
To: <sip:902@10.0.0.1>;tag=as2fb4ad13
Call-ID: b63e3b20c8fb2746@10.0.0.2
CSeq: 46026 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902@192.168.0.4>
Content-Type: application/sdp
Content-Length: 231
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@141.158.116.67>;tag=2235100740
To: <sip:902@141.158.116.67>;tag=as7f3a4393
Call-ID: 5D977977-C457-4D78-B1D3-84F994118AFF@217.137.52.48
CSeq: 15113 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:902@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
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum