Summary:ASTERISK-15313: chan sip removes peers like if srvlookup were active, but it is not
Reporter:Private Name (falves11)Labels:
Date Opened:2009-12-15 00:10:33.000-0600Date Closed:2009-12-17 10:28:40.000-0600
Versions:Frequency of
Description:These are my "sip show settings"
Global Settings:
 SIP Port:               5060
 Videosupport:           No                                                                                                                                                                        
 AutoCreatePeer:         Yes
 Allow unknown access:   Yes
 Allow subscriptions:    Yes
 Allow overlap dialing:  Yes
 Promsic. redir:         No
 SIP domain support:     No
 Call to non-local dom.: Yes
 URI user is phone no:   No
 Our auth realm          minixel.com
 Realm. auth:            No
 Always auth rejects:    No
 Call limit peers only:  No
 Direct RTP setup:       No
 User Agent:             Asterisk
 MWI checking interval:  10 secs
 Reg. context:           (not set)
 Caller ID:              0000000000
 From: Domain:          
 Record SIP history:     Off
 Call Events:            Off
 IP ToS SIP:             CS3
 IP ToS RTP audio:       EF
 IP ToS RTP video:       AF41
 T38 fax pt UDPTL:       Yes
 RFC2833 Compensation:   Yes
 SIP realtime:           Disabled

Global Signalling Settings:
 Codecs:                 0x105 (g723|ulaw|g729)
 Codec Order:            g729:20,ulaw:20,g723:30
 T1 minimum:             1500
 No premature media:     No
 Relax DTMF:             Yes
 Compact SIP headers:    No
 RTP Keepalive:          0 (Disabled)
 RTP Timeout:            30
 RTP Hold Timeout:       60
 MWI NOTIFY mime type:   text/plain
 DNS SRV lookup:         No
 Pedantic SIP support:   No
 Reg. min duration       60 secs
 Reg. max duration:      3600 secs                                                                                                                                                                
 Reg. default duration:  3600 secs
 Outbound reg. timeout:  20 secs
 Outbound reg. attempts: 0
 Notify ringing state:   Yes
 Notify hold state:      No
 SIP Transfer mode:      open
 Max Call Bitrate:       384 kbps
 Auto-Framing:           No

Default Settings:
 Context:                inbound
 Nat:                    Always                                                                                                                                                                    
 DTMF:                   auto
 Qualify:                0
 Use ClientCode:         No
 Progress inband:        Never
 Language:               en
 MOH Interpret:          default
 MOH Suggest:            
 Voice Mail Extension:   asterisk


But I get this error when reloading SIP:

host lookup failed for X.X.X.X., on peer Client.X.X.X.X., removing peer
Unable to lookup 'X.X.X.X.'
Comments:By: Leif Madsen (lmadsen) 2009-12-15 10:04:49.000-0600

You should know by now what information is required when reporting SIP issues:  sip history, sip debug, console output, relevant configuration information, etc... etc...

By: Private Name (falves11) 2009-12-15 10:10:28.000-0600

And I did. The only message that I get when I do a core set debug 10 is that one:

host lookup failed for X.X.X.X., on peer Client.X.X.X.X., removing peer
Unable to lookup 'X.X.X.X.'

and the output of the command "sip show settings" proves that DNS Srv lookup is disabled. I am at a  loss regarding what else do I need to supply at this point. The behavior is against the design. I think that a recent patch for dnssrv is responsible.
if you want access to my server, you are welcome.

By: Private Name (falves11) 2009-12-15 19:41:31.000-0600

I found the problem. My peer's host definition had an extra period at the right of the IP address. Nevertheless, Asterisk should not have attempted to loookup or validate the validity of the IP address, since it can be also a name that will be translated later, unless this is done by design.

Please advice.

By: Jason Parker (jparker) 2009-12-17 10:28:39.000-0600


Configuration error.