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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: "Weber Automundo" <sip:[EMAIL PROTECTED]>;tag=as4da46cda To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]:5060;user=phone;transport=udp>
set_destination: Parsing <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport
From: "Weber Automundo" <sip:[EMAIL PROTECTED]>;tag=as4da46cda
To: <sip:[EMAIL PROTECTED]>;tag=939809556
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED];user=phone>;tag=3858230914
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as601a996c
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> WWW-Authenticate: Digest realm="asterisk", nonce="0711b1d6" Content-Length: 0


to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' 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:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: <sip:[EMAIL PROTECTED]>;expires=120 Date: Wed, 16 Feb 2005 00:43:46 GMT Content-Length: 0


to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' 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:[EMAIL PROTECTED]>;tag=as59adf4c2 To: <sip:201.133.170.82> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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 '[EMAIL PROTECTED]'
set_destination: Parsing <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK2bdff4fa;rport
From: "Weber Automundo" <sip:[EMAIL PROTECTED]>;tag=as4da46cda
To: <sip:[EMAIL PROTECTED]>;tag=939809556
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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 '[EMAIL PROTECTED]'
Destroying call '[EMAIL PROTECTED]'
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:[EMAIL PROTECTED]>;tag=as370254a4
To: <sip:201.133.170.82>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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 '[EMAIL PROTECTED]'
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:[EMAIL PROTECTED];user=phone>;tag=3858230914
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as13999c1a
Call-ID: [EMAIL PROTECTED]
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as13999c1a Call-ID: [EMAIL PROTECTED] CSeq: 3 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> WWW-Authenticate: Digest realm="asterisk", nonce="33e2f5df" Content-Length: 0


to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' 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:[EMAIL PROTECTED];user=phone>;tag=3858230914 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as13999c1a Call-ID: [EMAIL PROTECTED] CSeq: 4 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED];user=phone>;tag=3858230914
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as13999c1a
Call-ID: [EMAIL PROTECTED]
CSeq: 4 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: <sip:[EMAIL PROTECTED]>;expires=120
Date: Wed, 16 Feb 2005 00:45:30 GMT
Content-Length: 0



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