| Summary: | ASTERISK-22870: dialplan entries pointing to SIP peers not defined in sip.conf just hangs the call | ||
| Reporter: | Arno Teigseth (arnotixe) | Labels: | |
| Date Opened: | 2013-11-20 08:41:20.000-0600 | Date Closed: | 2013-12-09 18:20:02.000-0600 |
| Priority: | Minor | Regression? | |
| Status: | Closed/Complete | Components: | Applications/app_dial |
| Versions: | 11.3.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | alpine linux 2.5 edge | Attachments: | ( 0) nonexistantSipbugfeature.txt |
| Description: | I had set up in extensions.conf
exten => 9904,1,Dial(SIP/930&SIP/arno) Usually when an extension isn't registered, it's just ignored when dialling 9904. Now, I deleted SIP peer "arno" from sip.conf and now dialling 9904 it just hangs. No audio, no ring on SIP peer 930 even if it is registered. To make things worse, if calling 9904 from an IAX2 trunk, the trunk hangs (see CLI verbosity 9 output attached). Only way to clear the condition is restarting the asterisk service. Which is bad. Of course, I should keep my pbx organized, so I don't know if this is a bug or a feature. But if there aren't negative side effects, wouldn't it be good if non-existant SIP peers were treated as non-registered ones? | ||
| Comments: | By: Arno Teigseth (arnotixe) 2013-11-20 08:42:18.240-0600 asterisk cli at core verbosity level 9 By: Arno Teigseth (arnotixe) 2013-11-20 08:44:52.392-0600 Tried also to see if the pbx tries to resolve SIP/arno to a hostname, and calling that hostname. Sniffing the network leads me to suspect that isn't the case, since I see no outgoing requests for hostname "arno". And adding a hostname "arno" to /etc/hosts yields no connection attepmts to the host. Seems to me as asterisk just gives up on finding SIP peer arno when it isn't defined in sip.conf, and abandons the call without hanging up, error code or anything. By: Rusty Newton (rnewton) 2013-11-21 15:29:56.834-0600 It is probably hung up in DNS resolution. A quick little test on my system showed that Asterisk does attempt to resolve the non-existing peer as a hostname. Although I don't get a hang of any kind. {noformat} -- Executing [6009@from-internal:1] Dial("SIP/6001-00000002", "SIP/barneyfief&SIP/6002,10") in new stack <snip> [Nov 21 15:18:42] DEBUG[21380][C-00000001]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'barneyfief' into... [Nov 21 15:18:42] DEBUG[21380][C-00000001]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'barneyfief' and port ''. [Nov 21 15:18:42] ERROR[21380][C-00000001]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("barneyfief", "(null)", ...): No address associated with hostname [Nov 21 15:18:42] WARNING[21380][C-00000001]: chan_sip.c:6201 create_addr: No such host: barneyfief {noformat} Do another test with DEBUG messages turned up to 5 in addition to your VERBOSE messages. (see logger.conf and check "logger show channels" on the CLI) You may find the issue there, but if not, attach the results to the issue. By: Rusty Newton (rnewton) 2013-12-09 18:19:52.689-0600 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 | ||