|Summary:||ASTERISK-17780: Hostname does not resolve when using realtime SIP|
|Reporter:||Jason Rogers (rogersja)||Labels:|
|Date Opened:||2011-05-02 14:51:49||Date Closed:||2011-07-26 14:48:25|
|Description:||Entering a hostname in the host field, does not resolve as it would if entered in sip.conf.|
This causes an incoming SIP URI call not to match to the peer and instead uses the default dialplan context, and configuration specified in the general section of sip.conf.
****** STEPS TO REPRODUCE ******
Enter domain name instead of IP address in host field in realtime sip table.
****** ADDITIONAL INFORMATION ******
If you substitute the FQDN for its IP in the database, everything works just fine and the incoming SIP URI call is recognized and matched to the proper peer.
|Comments:||By: Leif Madsen (lmadsen) 2011-05-05 09:03:29|
Please provide some debug logs (console output, etc...) so we can better understand what maybe going on. I've marked an issue as possible related.
By: Russell Bryant (russell) 2011-07-26 14:48:20.559-0500
Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines