| Summary: | ASTERISK-16770: Asterisk refuses to accept INVITE although auth info is correct | ||
| Reporter: | Stefan Tichy (st) | Labels: | |
| Date Opened: | 2010-10-05 07:22:08 | Date Closed: | 2011-06-07 14:05:21 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/DatabaseSupport |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | WARNING[16991] chan_sip.c: username mismatch, have <12>, digest has <13> WARNING[16991] chan_sip.c: username mismatch, have <12>, digest has <13> Asterisk 1.6.2.13, realtime SIP config, devices behind NAT (same IP), all phones are configured to use TLS. First phone does work Second phone cannot make outgoing calls and Asterisk logs that message Third phone does work (that is really strange) If first phone unregisters the second one will work until the first registers again. Two Snom 3x0 phones and one Nokia have been used, but this seems to be insignificant. ****** ADDITIONAL INFORMATION ****** NAT router is a debian lenny server using iptables. The problem is not related to bug 18048. Further information will be provided as soon as I find out more. | ||
| Comments: | By: Stefan Tichy (st) 2010-10-05 09:47:13 Did not found any problem if phones are configured to use UDP transport for SIP. By: Leif Madsen (lmadsen) 2010-10-05 15:36:08 Lets start with some logging such as configuration settings, SIP traces from the Asterisk console along with SIP history and debug level logging. By: Stefan Tichy (st) 2010-10-07 10:01:44 Currently it looks like the strange behavior of Asterisk is caused by a misconfiguration for some of the phones. I have no idea why this occurs only if transport tcp or tls is used. By: Leif Madsen (lmadsen) 2010-10-12 10:14:51 OK, so close this issue? By: Stefan Tichy (st) 2010-10-13 05:20:16 Yes, please close the issue. Asterisk behaves strange but until now I am not able to reproduce a problem using a clean configuration. | ||