[asterisk-bugs] [Asterisk 0011503]: Asterisk Realtime not resoving host names

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11503 
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Reported By:                akhan01
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11503
Category:                   Channels/chan_sip/DatabaseSupport
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.15  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             12-10-2007 10:58 CST
Last Modified:              12-14-2007 16:32 CST
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Summary:                    Asterisk Realtime not resoving host names
Description: 
Using asterisk Realtime I have noticed that incoming calls will not work
unless you change the HOST field from domain name to a physical IP
address.

If a domain name is specified for host the bellow NOTICE is seen on
asterisk
 NOTICE[3008] chan_sip.c: Call from '' to extension '13864345565' rejected
because extension not found.

We tested this many times and if a domain name is specified for HOST then
98%
of the incoming calls gets rejects with the above NOTICE.


====================================================================== 

---------------------------------------------------------------------- 
 akhan01 - 12-14-07 16:32  
---------------------------------------------------------------------- 
Bellow is the debug.
Notice "Found no matching peer or user for '64.34.181.47:5060'"

[Dec 14 16:14:59] VERBOSE[3008] logger.c: Using INVITE request as basis
request - 25cf9b160722c5a12f566a3870da13e1 at 64.34.181.47
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Found no matching peer or user
for '64.34.181.47:5060'
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Found RTP audio format 0
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Found RTP audio format 101
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Peer audio RTP is at port
64.34.181.47:15326
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Found audio description format
PCMU for ID 0
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Found audio description format
telephone-event for ID 101
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Capabilities: us - 0x4 (ulaw),
peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Non-codec capabilities (dtmf):
us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Peer audio RTP is at port
64.34.181.47:15326
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Looking for 17058124935 in
default (domain 75.126.42.12)
[Dec 14 16:14:59] VERBOSE[3008] logger.c:
<--- Reliably Transmitting (NAT) to 64.34.181.47:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
64.34.181.47:5060;branch=z9hG4bK1c911328;received=64.34.181.47;rport=5060
From: "NOT FOUND" <sip:19544939151 at 64.34.181.47>;tag=as2a8db2da
To: <sip:17058124935 at 75.126.42.12>;tag=as058e2267
Call-ID: 25cf9b160722c5a12f566a3870da13e1 at 64.34.181.47
CSeq: 102 INVITE
User-Agent: Timeous PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
[Dec 14 16:14:59] NOTICE[3008] chan_sip.c: Call from '' to extension
'17058124935' rejected because extension not found.
[Dec 14 16:14:59] VERBOSE[3008] logger.c: Scheduling destruction of SIP
dialog '25cf9b160722c5a12f566a3870da13e1 at 64.34.181.47' in 32000 ms (Method:
INVITE)
[Dec 14 16:14:59] VERBOSE[3008] logger.c: 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
12-14-07 16:32  akhan01        Note Added: 0075435                          
======================================================================