[Asterisk-Users] SIP call troubleshooting

[Marko Rakar]

Can someone help me what went wrong with this call?

This call was initiated from dev/ttyI0 device on my asterisk server to
mediatrix unit. Mediatrix unit user received the call and call started.
I can hear them OK but they can not hear me correctly (cut-off sound,
noise). Call was finally hunged up.

Can anyone point out if there was something wrong?



-*CLI> sip debug
SIP Debugging Enabled
Asterisk Ready.
We're at 192.168.3.6 port 12556
Answering/Requesting with preferred capability 8
Answering/Requesting with preferred capability 4
12 headers, 9 lines
Reliably Transmitting:
INVITE sip:304 at 192.168.3.211 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
From: "0" <sip:0 at 192.168.3.6>;tag=as1dbb6ad3
To: <sip:304 at 192.168.3.211>
Contact: <sip:0 at 192.168.3.6>
Call-ID: 70367b6e2c6e4f4c41cf74e1356e5f77 at 192.168.3.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 02 Apr 2004 12:01:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 182

v=0
o=root 10202 10202 IN IP4 192.168.3.6
s=session
c=IN IP4 192.168.3.6
t=0 0
m=audio 12556 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
 (no NAT) to 192.168.3.211:5060
-*CLI>

Sip read:
SIP/2.0 180 Ringing
Call-ID: 70367b6e2c6e4f4c41cf74e1356e5f77 at 192.168.3.6
CSeq: 102 INVITE
From: 0 <sip:0 at 192.168.3.6>;tag=as1dbb6ad3
To: <sip:304 at 192.168.3.211>;tag=acc03844-c7bb79c5
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
Content-Length: 0


7 headers, 0 lines
-*CLI>

Sip read:
SIP/2.0 200 OK
Call-ID: 70367b6e2c6e4f4c41cf74e1356e5f77 at 192.168.3.6
CSeq: 102 INVITE
From: 0 <sip:0 at 192.168.3.6>;tag=as1dbb6ad3
To: <sip:304 at 192.168.3.211>;tag=acc03844-c7bb79c5
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
Content-Length: 152
Content-Type: application/sdp
Contact: 304 <sip:304 at 192.168.3.211:5060>
Allow: INVITE, ACK, BYE, CANCEL, REFER

v=0
o=MxSIP 0 0 IN IP4 192.168.3.211
s=SIP Call
c=IN IP4 192.168.3.211
t=0 0
m=audio 5004 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000

10 headers, 8 lines
Found audio format ALAW
Found audio format UNKN
Found description format PCMA
Found description format PCMU
Capabilities: us - 12, them - 12/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:304 at 192.168.3.211:5060>
set_destination: Parsing <sip:304 at 192.168.3.211:5060> for address/port
to send to
set_destination: set destination to 192.168.3.211, port 5060
Transmitting:
ACK sip:304 at 192.168.3.211:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
From: "0" <sip:0 at 192.168.3.6>;tag=as1dbb6ad3
To: <sip:304 at 192.168.3.211>;tag=acc03844-c7bb79c5
Contact: <sip:0 at 192.168.3.6>
Call-ID: 70367b6e2c6e4f4c41cf74e1356e5f77 at 192.168.3.6
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.3.211:5060
set_destination: Parsing <sip:304 at 192.168.3.211:5060> for address/port
to send to
set_destination: set destination to 192.168.3.211, port 5060
Reliably Transmitting:
BYE sip:304 at 192.168.3.211:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
From: "0" <sip:0 at 192.168.3.6>;tag=as1dbb6ad3
To: <sip:304 at 192.168.3.211>;tag=acc03844-c7bb79c5
Contact: <sip:0 at 192.168.3.6>
Call-ID: 70367b6e2c6e4f4c41cf74e1356e5f77 at 192.168.3.6
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 192.168.3.211:5060
-*CLI>

Sip read:
SIP/2.0 200 OK
Call-ID: 70367b6e2c6e4f4c41cf74e1356e5f77 at 192.168.3.6
CSeq: 103 BYE
From: 0 <sip:0 at 192.168.3.6>;tag=as1dbb6ad3
To: <sip:304 at 192.168.3.211>;tag=acc03844-c7bb79c5
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
Content-Length: 0


7 headers, 0 lines
-*CLI>


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mailto:marko at printel.hr
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