[Asterisk-Users] Help Please!!!!

Wed Feb 16 12:33:48 MST 2005

I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is 
that one of them is dropping calls an I can't figure out what is the 
problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem.

Any help will be appreciate

Thanks

Erick Weber

VoIP*CLI> sip debug peer 1088
SIP Debugging Enabled for IP: 201.133.170.82:5060
Peer RTP is at port 192.168.1.69:0
Peer RTP is at port 192.168.1.69:0
    -- Executing Dial("SIP/404-cbc9", "SIP/1088|60|tr") in new stack
We're at XXX.XXX.XXX.XXX port 17506
Answering/Requesting with root capability 256
12 headers, 8 lines
Reliably Transmitting:
INVITE sip:1088 at 201.133.170.82 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport
From: "Weber Automundo" <sip:404 at XXX.XXX.XXX.XXX>;tag=as4da46cda
To: <sip:1088 at 201.133.170.82>
Contact: <sip:404 at XXX.XXX.XXX.XXX>
Call-ID: 00b325641a0f0d680014aad165ce6d4c at XXX.XXX.XXX.XXX
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 16 Feb 2005 00:43:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 164

v=0
o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX
s=session
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 17506 RTP/AVP 18
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
 (NAT) to 201.133.170.82:5060
    -- Called 1088
    -- SIP/1088-ec82 is ringing
Found RTP audio format 18
Found RTP audio format 101
Peer RTP is at port 192.168.1.2:0
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100(G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), 
combined - 0x100(G729A)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 
0x1(G723)
list_route: hop: <sip:1088 at 192.168.1.2:5060;user=phone;transport=udp>
set_destination: Parsing 
<sip:1088 at 192.168.1.2:5060;user=phone;transport=udp> for address/port to 
send to
set_destination: set destination to 192.168.1.2, port 5060
Transmitting:
ACK sip:1088 at 192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport
From: "Weber Automundo" <sip:404 at XXX.XXX.XXX.XXX>;tag=as4da46cda
To: <sip:1088 at 201.133.170.82>;tag=939809556
Contact: <sip:404 at XXX.XXX.XXX.XXX>
Call-ID: 00b325641a0f0d680014aad165ce6d4c at XXX.XXX.XXX.XXX
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 201.133.170.82:5060
    -- SIP/1088-ec82 answered SIP/404-cbc9
    -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82
    -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82
Using latest request as basis request
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as601a996c
Call-ID: 30194281 at 192.168.1.2
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1088 at XXX.XXX.XXX.XXX>
Content-Length: 0

 to 201.133.170.82:5060
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as601a996c
Call-ID: 30194281 at 192.168.1.2
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1088 at XXX.XXX.XXX.XXX>
WWW-Authenticate: Digest realm="asterisk", nonce="0711b1d6"
Content-Length: 0

 to 201.133.170.82:5060
Scheduling destruction of call '30194281 at 192.168.1.2' in 15000 ms
Using latest request as basis request
Sending to 192.168.1.2 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as601a996c
Call-ID: 30194281 at 192.168.1.2
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1088 at XXX.XXX.XXX.XXX>
Content-Length: 0

 to 201.133.170.82:5060
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as601a996c
Call-ID: 30194281 at 192.168.1.2
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: <sip:1088 at XXX.XXX.XXX.XXX>;expires=120
Date: Wed, 16 Feb 2005 00:43:46 GMT
Content-Length: 0

 to 201.133.170.82:5060
Scheduling destruction of call '30194281 at 192.168.1.2' in 15000 ms
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:201.133.170.82 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK0972cae7
From: "asterisk" <sip:asterisk at XXX.XXX.XXX.XXX>;tag=as59adf4c2
To: <sip:201.133.170.82>
Contact: <sip:asterisk at XXX.XXX.XXX.XXX>
Call-ID: 00373ed80ed1caa4513efc600569cc96 at XXX.XXX.XXX.XXX
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 16 Feb 2005 00:43:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 201.133.170.82:5060
Destroying call '00373ed80ed1caa4513efc600569cc96 at XXX.XXX.XXX.XXX'
set_destination: Parsing 
<sip:1088 at 192.168.1.2:5060;user=phone;transport=udp> for address/port to 
send to
set_destination: set destination to 192.168.1.2, port 5060
Reliably Transmitting:
BYE sip:1088 at 192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2bdff4fa;rport
From: "Weber Automundo" <sip:404 at XXX.XXX.XXX.XXX>;tag=as4da46cda
To: <sip:1088 at 201.133.170.82>;tag=939809556
Contact: <sip:404 at XXX.XXX.XXX.XXX>
Call-ID: 00b325641a0f0d680014aad165ce6d4c at XXX.XXX.XXX.XXX
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 201.133.170.82:5060
  == Spawn extension (hi, 1088, 1) exited non-zero on 'SIP/404-cbc9'
    -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 
192.168.1.2
Destroying call '00b325641a0f0d680014aad165ce6d4c at XXX.XXX.XXX.XXX'
Destroying call '30194281 at 192.168.1.2'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:201.133.170.82 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK0689fc21
From: "asterisk" <sip:asterisk at XXX.XXX.XXX.XXX>;tag=as370254a4
To: <sip:201.133.170.82>
Contact: <sip:asterisk at XXX.XXX.XXX.XXX>
Call-ID: 78dea4a01158e0154b01eba37450a08b at XXX.XXX.XXX.XXX
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 16 Feb 2005 00:44:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 201.133.170.82:5060
Destroying call '78dea4a01158e0154b01eba37450a08b at XXX.XXX.XXX.XXX'
Using latest request as basis request
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as13999c1a
Call-ID: 30194281 at 192.168.1.2
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1088 at XXX.XXX.XXX.XXX>
Content-Length: 0

 to 201.133.170.82:5060
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as13999c1a
Call-ID: 30194281 at 192.168.1.2
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1088 at XXX.XXX.XXX.XXX>
WWW-Authenticate: Digest realm="asterisk", nonce="33e2f5df"
Content-Length: 0

 to 201.133.170.82:5060
Scheduling destruction of call '30194281 at 192.168.1.2' in 15000 ms
Using latest request as basis request
Sending to 192.168.1.2 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as13999c1a
Call-ID: 30194281 at 192.168.1.2
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1088 at XXX.XXX.XXX.XXX>
Content-Length: 0

 to 201.133.170.82:5060
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060
From: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=3858230914
To: <sip:1088 at XXX.XXX.XXX.XXX;user=phone>;tag=as13999c1a
Call-ID: 30194281 at 192.168.1.2
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: <sip:1088 at XXX.XXX.XXX.XXX>;expires=120
Date: Wed, 16 Feb 2005 00:45:30 GMT
Content-Length: 0 