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 00:10:33.000-0600 | Date Closed: | 2009-12-17 10:28:40.000-0600 |
Priority: | Trivial | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | These are my "sip show settings" Global Settings: ---------------- SIP Port: 5060 Bindaddress: 0.0.0.0 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 ****** ADDITIONAL INFORMATION ****** 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 Closing. Configuration error. |