[asterisk-bugs] [Asterisk 0019252]: error on SIP INVITE when host's external IP changes

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=19252 
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Reported By:                Newborn
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19252
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.2.18 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 duplicate
Duplicate:                  0
Fixed in Version:           
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Date Submitted:             2011-05-09 08:58 CDT
Last Modified:              2011-05-10 17:14 CDT
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Summary:                    error on SIP INVITE when host's external IP changes
Description: 
I have got 2 Asterisk boxes connected to each other via Internet (both are
behind NAT). One have static IP (A), other is dynamic (B). Dynamic
registers to static respectively. Other client C is connected to B and
placing a call via B to A.  So when public IP of host B changes, I cannot
place calls using client C, from B to A, until Asterisk A is being
rebooted. dnsmgr is enabled. when i tracert to host B, i see the correct IP
address, as above in SIP headers.

some logs...

<--- SIP read from UDP:91.77.113.139:5060 --->
INVITE sip:0100@***.***.***.*** SIP/2.0
Via: SIP/2.0/UDP 91.77.113.139:5060;branch=z9hG4bK3b9d829a;rport
Max-Forwards: 70
From: "6001" <sip:6001 at 91.77.113.139>;tag=as1c3b8a40
To: <sip:0100@***.***.***.***>
Contact: <sip:6001 at 91.77.113.139>
Call-ID: 7a50c20b4e7b47416716ba90032a1ba2 at 91.77.113.139
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.18
Date: Mon, 09 May 2011 13:55:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1783255373 1783255373 IN IP4 91.77.113.139
s=Asterisk PBX 1.6.2.18
c=IN IP4 91.77.113.139
t=0 0
m=audio 23832 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->

[May  9 17:40:45] Using INVITE request as basis request -
6f1763b03df75d2909c749c53f7939fe at 91.77.155.19
[May  9 17:40:45] No matching peer for '6001' from '91.77.155.19:5060'
[May  9 17:40:45] Found RTP audio format 8
[May  9 17:40:45] Found RTP audio format 101
[May  9 17:40:45] Found audio description format PCMA for ID 8
[May  9 17:40:45] Found audio description format telephone-event for ID
101
[May  9 17:40:45] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer -
audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8
(alaw)
[May  9 17:40:45] Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
[May  9 17:40:45] Peer audio RTP is at port 91.77.155.19:22076
[May  9 17:40:45] Looking for 0100 in default (domain ***.***.***.***)
[May  9 17:40:45]
<--- Reliably Transmitting (NAT) to 91.77.155.19:5060 --->
SIP/2.0 404 Not Found


6001 is client C peer.
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Relationships       ID      Summary
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duplicate of        0018662 Incoming calls stop working after chang...
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Issue History 
Date Modified    Username       Field                    Change               
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2011-05-10 17:14 lmadsen        Status                   feedback => closed  
2011-05-10 17:14 lmadsen        Resolution               open => duplicate   
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