| 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 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/DatabaseSupport |
| Versions: | 1.8.4 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ||
| 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 | ||