Summary:	ASTERISK-14151: SIP stops working with multiple REGISTER statements in sip.conf
Reporter:	philipp2 (philipp2)	Labels:
Date Opened:	2009-05-17 08:58:33	Date Closed:	2011-06-07 14:01:03
Priority:	Blocker	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/Registration
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	System 1 (1.4.18, 2.8 GHz) with "register =>" statements in sip.conf, one of them with sipgate.de. The SIP part of the system becomes unresponsive after a couple of days. Removing the second "register =>" for sipgate.de made the system stable. System 2 (1.6 GHz, formerly 1.4.21 and now 1.4.24.1): Three "register =>" statements to the SAME registrar (also Asterisk, btw). After a SIP RELOAD the entire SIP part of the system becomes unresponsive, meaning that also local LAN phones cannot register anymore. SIP SHOW REGISTRY shows "Request Sent" (and sometimes "Unregistered" for one of the three). Reducing from 3 to 2 "register =>" statemens sometimes make things work, reducing from 3 to 1 heals it completely. Moreover: After the SIP RELOAD is issued on the CLI there is no further output of any other CLI command, however with EXIT and again an "asterisk -r" CLI output can be re-gained. ****** ADDITIONAL INFORMATION ****** OS: CentOS 5 (PIAF), Debian Etch Witnessed in Asterisk versions: 1.4.18, 1.4.21, 1.4.24.1 Note: This form does not permit me to select 1.4.24.1 as Asterisk version. In both cases Asterisk is placed behind NAT using "externip=" in sip.conf. Realtime is NOT involved, configuration by flat files only, so this is probably related to ASTERISK-12245 but NOT exactly the same problem. Please note that these are both prodcution system where I will unfortunately not be able to run tests with SVN.
Comments:	By: philipp2 (philipp2) 2009-05-17 09:02:44 Please excuse the typos in "statemenTs" and "prodUCtion" and "systemS". :-) By: philipp2 (philipp2) 2009-05-17 09:07:00 Additional Info: In the faulty state SIP DEBUG shows that Asterisk does not receive any responses to its requests for REGISTRATION. By: philipp2 (philipp2) 2009-05-17 09:43:17 Possibly related: https://issues.asterisk.org/view.php?id=15052 tcpdump shows that in fact we are receiving registration responses ("200 OK" and "401 Unauthorized", or "100 Trying" and "401 Unauthorized") from the remote registrar, however "SIP SET DEBUG IP xxx" only reveals never-ending outbound "Retransmitting #x (no NAT)" messages but no inbound SIP messages at all. So it appears that Asterisk has stopped listening. Please note that this issue is not 100% reproducible, sometimes it is necessary to add a 4th or 5th "register =>" statement or issue SIP RELOAD twice. The register works by itself, it does not refer to a [section] in sip.conf. Retransmitting #2 (no NAT) to a.b.c.d:5060: REGISTER sip:a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK0c9d8347;rport From: <sip:4321@a.b.c.d>;tag=as1359ea00 To: <sip:4321@a.b.c.d> Call-ID: 36233f5a4545d8076b76af432e0437ca@127.0.0.1 CSeq: 122 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:123@192.168.1.20> Event: registration Content-Length: 0 BTW: Shouldn't we see our own external IP address here, as set with externip? By: John S. (johnakabean) 2009-05-17 22:15:55 Multiple submissions reported of this: https://issues.asterisk.org/view.php?id=15052 By: Leif Madsen (lmadsen) 2009-06-16 13:37:00 Duplicate of 15052. Please track this issue in that report. Thanks! By: Leif Madsen (lmadsen) 2009-06-25 07:49:24 Apparently this issue differs slightly from that it is related to, so I'm reopening this issue to be tracked separately. By: Federico Santulli (fsantulli) 2009-06-25 08:07:37 This bug is surely a DNS related problem, if you look over /main/dns.c you'll find: "Asterisk DNS is synchronus at this time. This means that if your DNS does not work properly, Asterisk might not start properly or a channel may lock." Try to change nameservers in /etc/resolv.conf. By: Leif Madsen (lmadsen) 2009-06-29 11:23:45 I'm closing this per the note by fsantulli. If someone thinks this really needs to be open, and is an issue here, please find me on #asterisk-bugs IRC channel as 'leifmadsen'. Thanks!