How to allow registered sip users to call without re-authenticationinsecure =yes/very are deprecated in 1.8I want to avoid fromuser= in peer configuration. When I add this in peer asterisk, my asterisk accepts call otherwise it says username mismatch.Ple..
On 01/05/12 16:42, Joseph wrote: >I just noticed after upgrade from Asterisk 1.4.39 to 1.8.8 >my caller ID is not working > >WARNING: chan_sip.c:13956 check_auth: username mismatch, have <11>, digest has >NOTICE: chan_sip.c:22048 handle_request_invi..
In the thread Interesting attack tonight & fail2ban them Bruce B mentioned it would be nice to have input from the Community to come up with the best set of fail2ban filters. Thats a great idea. So lets start with Bruces filters (thanks!) and take..
Im getting various codec related warnings after upgrading to 1.8.Did I miss something in the UPGRADE file?Does Asterisk no longer transcode 8-)? WARNING: channel.c:4909 ast_write: Codec mismatch on channel DAHDI/i1/12124221200-74 setting wr..
Asterisk seems to try to authenticate incoming INVITE based on the [section] in sip.conf and not the username specified. I just removed the insecure option from my sip.conf requesting every connection to be authenticated. I added the match_auth_username=..
Dan Journowrote: >> Ive got a Polycom 501 that I just cant seem to get lines 2 and 3 to work on. >>Line 1 works fine. > > Last time I had that issue, it resolved itself when i restarted Asterisk. Any ideas as to what causes it, though? > Are you a..
2011/5/5 Richard Kenner > I recently tried to update my Aastra 57i to version 3.2 and ran into > a problem.It wont properly register and says contact mismatch. > I added sip contact matching: 2 to aastra.cfg, but that didnt help. > > When I look at ..
Look like codec mismatch..
I have my asterisk Server A registered as a client with another asterisk Server B. When I place a call from Server A to B I get the following: WARNING: chan_sip.c:12673 check_auth: username mismatch, have , digest has NOTICE: chan_sip.c:19..