| Summary: | ASTERISK-14434: rtpkeepalive option doesn't always work | ||
| Reporter: | Stanislaw Pitucha (viraptor) | Labels: | |
| Date Opened: | 2009-07-08 10:05:09 | Date Closed: | 2011-06-07 14:08:25 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | I've tried enabling rtpkeepalive option (2 seconds) in a server without nat or any specific dialplan (just a Dial(SIP/...) ). It works fine on one server, but doesn't on the other. I cannot find any reason for it though. Configs are almost the same. The only real difference between them is that one is multihomed (the one where rtpkeepalive fails), but I'm not sure that's related at all. All audio passes through asterisk in both setups. I expect the rtpkeepalive to be sent while the call is on hold. How can I debug this issue? | ||
| Comments: | By: Olle Johansson (oej) 2009-08-26 07:25:18 Does "rtp debug" show the keepalives? By: Stanislaw Pitucha (viraptor) 2009-08-26 07:28:21 No, it doesn't. Listing the sip option confirms that it's enabled though. By: Leif Madsen (lmadsen) 2009-09-18 09:25:50 I'm going to close this issue for now because this feels more like a support issue (or at least the very early stages of bug triage) where it would make more sense to get help and information from the asterisk-users mailing list. If you gather enough information to determine where (or if) the bug exists, then please feel free to reopen this issue. Thanks! | ||