Summary: | ASTERISK-13865: SIP channels never close when using tcp or tls | ||
Reporter: | David Vossel (dvossel) | Labels: | |
Date Opened: | 2009-09-14 14:09:16 | Date Closed: | 2009-09-17 09:51:57 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/TCP-TLS |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | When making a call via tcp or tls from one Asterisk box to another, the channel on the receiving box never goes away. If I change the peers to use udp this does not happen. The channel starts out looking something like this. 10.24.18.230 asterisk 7019c6102fbff27 0x4 (ulaw) No Rx: INVITE Then after the call is over, the format changes to (nothing) and it never goes away. It is as if the channel never gets hung up. 10.24.18.230 asterisk 7019c6102fbff27 0x0 (nothing) No Rx: INVITE ---dialplan logic for the extension being called--- exten => 5200,1, Answer() exten => 5200,n, PlayBack(tt-weasels) exten => 5200,n, hangup() ****** ADDITIONAL INFORMATION ****** *CLI> -- Executing [5200@local:1] Answer("SIP/SIP_manbearpig-0a35f570", "") in new stack -- Executing [5200@local:2] Playback("SIP/SIP_manbearpig-0a35f570", "tt-weasels") in new stack -- <SIP/SIP_manbearpig-0a35f570> Playing 'tt-weasels.ulaw' (language 'en') -- Executing [5200@local:3] Hangup("SIP/SIP_manbearpig-0a35f570", "") in new stack == Spawn extension (local, 5200, 3) exited non-zero on 'SIP/SIP_manbearpig-0a35f570' *CLI> *CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry xxx.xxx.xxx.xxx asterisk 31146b7d629a4ba 0x0 (nothing) No Rx: INVITE 1 active SIP dialog *CLI> -- Executing [5200@local:1] Answer("SIP/SIP_manbearpig-0a369158", "") in new stack -- Executing [5200@local:2] Playback("SIP/SIP_manbearpig-0a369158", "tt-weasels") in new stack -- <SIP/SIP_manbearpig-0a369158> Playing 'tt-weasels.ulaw' (language 'en') sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry xxx.xxx.xxx.xxx asterisk 31146b7d629a4ba 0x0 (nothing) No Rx: INVITE xxx.xxx.xxx.xxx asterisk 7019c6102fbff27 0x4 (ulaw) No Rx: INVITE 2 active SIP dialogs *CLI> -- Executing [5200@local:3] Hangup("SIP/SIP_manbearpig-0a369158", "") in new stack == Spawn extension (local, 5200, 3) exited non-zero on 'SIP/SIP_manbearpig-0a369158' sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry xxx.xxx.xxx.xxx asterisk 31146b7d629a4ba 0x0 (nothing) No Rx: INVITE xxx.xxx.xxx.xxx asterisk 7019c6102fbff27 0x0 (nothing) No Rx: INVITE 2 active SIP dialogs | ||
Comments: | By: Elazar Broad (ebroad) 2009-09-16 21:08:02 dvossel, I wonder if this is related to https://issues.asterisk.org/view.php?id=15896. By: David Vossel (dvossel) 2009-09-17 09:51:56 file resolved this issue in issue ASTERISK-14830. ebroad tested and verified file's commit fixes this as well. |