| Summary: | ASTERISK-20539: When the peer for an outbound call is defined by a round-robin DNS, the response to the SIP 407 challenge goes to a different server than the original SIP INVITE (and from which the 407 came) | ||
| Reporter: | Norman Henderson (nhenderson) | Labels: | |
| Date Opened: | 2012-10-09 09:16:49 | Date Closed: | 2012-10-11 16:21:31 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | 1.8.11.1 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Asterisk 1.8.11.1-1digium1~lucid built by pbuilder @ nighthawk on a x86_64 running Linux on 2012-04-24 15:31:40 UTC Linux VM-CEML 2.6.32-43-server #97-Ubuntu SMP Wed Sep 5 16:56:41 UTC 2012 x86_64 GNU/Linux - Ubuntu 10.04.4 LTS - fully updated FreePBX 2.10.1.2 - fully updated | Attachments: | ( 0) extract |
| Description: | outbound.vitelity.net is a round-robin DNS name with several servers. The SIP trunk is defined as: [vitel-outfail] disallow=all username=username type=peer trustrpid=yes tos=ef sendrpid=yes secret=secret qualify=yes insecure=port,invite host=outbound.vitelity.net fromuser=username dtmfmode=auto canreinvite=no allow=g729 context=from-trunk-sip-vitel-outfail See debug file: Line 1321: Asterisk sends INVITE to the first address it got from the DNS lookup, 64.2.142.190. Line 1360: the provider responds from the same IP with 407 Proxy Authentication Required Line 1391ff: Asterisk does a new DNS lookup, gets a different address 64.2.142.188, and sends the digest credentials there. The provider says this is non-RFC-compliant (and indeed doesn't make much sense). Their server sends back another 407 from 64.2.142.188 and the cycle repeats a few times before Asterisk gives up. There is a mystery in that the system used to work (or perhaps the problem was just more intermittent and blamed on a bad 'net connection). I am not aware of changes to my system and the provider says his didn't change either during the time window this went from working (usually at least) to consistently failing. | ||
| Comments: | By: Norman Henderson (nhenderson) 2012-10-09 09:19:20.565-0500 core set debug 5; sip set debug on my user, IP, DNS and tel# have been globally altered for security reasons By: Michael L. Young (elguero) 2012-10-09 22:55:23.980-0500 Can you try to reproduce with a recent version of Asterisk 1.8.x? 1.8.17 is the latest version and in looking through the ChangeLog, I see some bug fixes around this area of the code. Worth a shot since your bug might already have been fixed. By: Norman Henderson (nhenderson) 2012-10-10 01:20:39.724-0500 Right - good suggestion. However it highlights "the problem with Linux" - 1.8.11.1 is the latest package for Lucid, Digium doesn't offer a package at all for Precise which is my next planned OS - Ubuntu offers only as recent as 1.8.13.1 for Precise. Building from source throws off the whole system maintenance strategy (plus I'm not that good at it and usually it turns into a mess). Anyway this is on a VM so I'll see about taking a snapshot and having a whirl at the 1.8.17 source in a bit. By: Norman Henderson (nhenderson) 2012-10-11 15:00:28.622-0500 After an upgrade to Ubuntu 12.04 and to Asterisk 1.8.17.0 (built from source), this problem appears to be fixed. There is no audio, but that's a different problem and requires more investigation before being a bug. It would be good to get a more recent .deb out there... Other than that, from my viewpoint this can be closed. Thanks for the help! By: Michael L. Young (elguero) 2012-10-11 16:21:31.197-0500 Closing per reporter's request. This bug has been fixed in the latest version of Asterisk. | ||