[asterisk-users] Fw: Sip Hangup after critical packet SIP DEBUG attached

[Zakir Mahomedy]

HI
 
I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn)
 
Out going calls from asterisk to the ata works fine
Incoming calls from the ata to asterisk cuts off with the error msg
 
Maximum retries exceeded on transmission 70854efe-4157e3a8 at 10.168.7.103 for 
seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Dec  6 13:52:43] WARNING[3921]: chan_sip.c:3858 retrans_pkt:
 Hanging up call 70854efe-4157e3a8 at 10.168.7.103 - no reply to our critical 
packet (see doc/sip-retransmit.txt).

I been googling this error and it was mentioned to use
t1min= 500 however its only delaying the problem.
 
any ideas on what is the cause of this problem.
Only 2-3 atas are having this problem the rest are fine.
 
Here is the sip debug
the sip invites are not being received
and in one of the message a busy response was sent back.
 Retransmitting #4 (no NAT) to 10.168.7.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103
From: Ridge <sip:287 at 10.10.0.1>;tag=f314fd35733eba9bo0
To: <sip:204 at 10.10.0.1>;tag=as4593172b
Call-ID: f54a1cbd-891ce0b3 at 10.168.7.103
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:204 at 41.146.208.131>
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 777980638 777980638 IN IP4 41.146.208.131
s=Asterisk PBX 1.6.2.13
c=IN IP4 41.146.208.131
t=0 0
m=audio 19726 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
 
Retransmitting #5 (no NAT) to 10.168.7.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103
From: Ridge <sip:287 at 10.10.0.1>;tag=f314fd35733eba9bo0
To: <sip:204 at 10.10.0.1>;tag=as4593172b
Call-ID: f54a1cbd-891ce0b3 at 10.168.7.103
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:204 at 41.146.208.131>
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 777980638 777980638 IN IP4 41.146.208.131
s=Asterisk PBX 1.6.2.13
c=IN IP4 41.146.208.131
t=0 0
m=audio 19726 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Reliably Transmitting (no NAT) to 10.168.7.103:5060:
OPTIONS sip:287 at 10.168.7.103:5060 SIP/2.0
Via: SIP/2.0/UDP 41.146.208.131:5060;branch=z9hG4bK226c4b89;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 41.146.208.131>;tag=as21bdce7e
To: <sip:287 at 10.168.7.103:5060>
Contact: <sip:asterisk at 41.146.208.131>
Call-ID: 062f8f6c4e7f5929487f3db12a93f7c2 at 41.146.208.131
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 06 Dec 2010 12:47:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
　
---
<--- SIP read from UDP:10.168.7.103:5060 --->
SIP/2.0 486 Busy Here
To: <sip:287 at 10.168.7.103:5060>;tag=18c8b9ab85ca5068i0
From: "asterisk" <sip:asterisk at 41.146.208.131>;tag=as21bdce7e
Call-ID: 062f8f6c4e7f5929487f3db12a93f7c2 at 41.146.208.131
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 41.146.208.131:5060;branch=z9hG4bK226c4b89
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
　
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '062f8f6c4e7f5929487f3db12a93f7c2 at 41.146.208.131' 
Method: OPTIONS
Retransmitting #6 (no NAT) to 10.168.7.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103
From: Ridge <sip:287 at 10.10.0.1>;tag=f314fd35733eba9bo0
To: <sip:204 at 10.10.0.1>;tag=as4593172b
Call-ID: f54a1cbd-891ce0b3 at 10.168.7.103
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:204 at 41.146.208.131>
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 777980638 777980638 IN IP4 41.146.208.131
s=Asterisk PBX 1.6.2.13
c=IN IP4 41.146.208.131
t=0 0
m=audio 19726 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
 
 
zakir
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