The process for upgrading would greatly depend on how Asterisk was installed in the first place.
If Asterisk was installed from source, then a fresh download of source followed by the usual configure/make/etc commands would do the trick. PaulH Veselin K wrote: > Hello Paul, > thanks for the reply. > > Could you please tell me what is the process called so I can > research it further. > > > > Thank you. > > Veselin K > > On Mon, Nov 17, 2008 at 10:47:47AM +1100, Paul Hales wrote: > >> This process has been greatly improved in the latest versions of >> Asterisk - might be time to upgrade. >> >> PaulH >> >> >> [EMAIL PROTECTED] wrote: >> >>> Hello, >>> I'm running an Asterisk 1.4.14 on a linux machine. >>> Serving SIP Snom users. >>> >>> I've noticed that each time Asterisk is restarted, for the first 5-10 >>> minutes, the SIP users can dial but cannot be dialed until each phone >>> re-registers itself against the server. >>> >>> So only after the "Saved useragent...for peer 111" line appears on the >>> Asterisk console, then the 111 user can be reached. >>> >>> What exactly is this process? >>> >>> Is it that the phones send their extension/password details to the >>> server at specific intervals or does the server send a broadcast >>> message, looking for phones? >>> >>> Is there any way to cache/save this SIP useragent information so in case >>> the server is restarted, the user need not wait for their phone to >>> re-register? >>> >>> Also I believe that it is sufficient for the user to just pickup their >>> handset in order to force their phone to re-register quicker. >>> >>> However I'd like to avoid asking the users to do that. >>> >>> Thank you much. >>> >>> Veselin K >>> >>> >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
