OK, read all about the patch, thanks for the fix Richard. I would like to apply this patch to my current 1.8.7.1 but I am afraid I don't have a clue how.
Is this just a case of getting a copy of app_followme.c and replacing it on my current Asterisk install? If not, do I have to grab a new Asterisk version and recompile everything and start over? I know I can save my configs but I was hoping for a simple fix without having to recompile Asterisk from source etc. Thanks for any help. --Todd On Thu, Dec 15, 2011 at 9:07 PM, Todd Routhier <[email protected]> wrote: > No, I get no error in the CLI at all, just shows that the followme is > being executed then dumps straight to Vmail which is defined in my dialplan > on the next line after calling the followme. > > I checked out the link and it also shows problems with callerid not > passing, this is also a problem for me and that was what I was going to > tackle next. > > I will checkout the patch, I have never applied a patch though, only done > fresh installs. So, I will need to figure that out. > > I am running Asterisk 1.8.7.1 to be more specific. > > Thanks Richard. > > > On Thu, Dec 15, 2011 at 8:56 PM, Richard Mudgett <[email protected]>wrote: > >> > ***************** >> > Summary: >> > >> > >> > I need to be able to ring multiple numbers in followme.conf at the >> > same time, even if one of the SIP extensions is unreachable. >> > This works in 1.4.8 but not in 1.8, just barfs and sends to voice >> > mail instead of ringing the other 2 extensions on the same line in >> > the followme.conf >> > >> > >> > See more details below. >> > ***************** >> > >> > I decided to mess around with followme and it actually suits my needs >> > quit well. I want to know what number the caller called into when my >> > cell phone rings, then decide if I want to answer it by pressing 1 >> > or not. Also helps with making sure voicemail is only left on my >> > Asterisk voicemail instead of my cell phone voice mail. >> > >> > >> > So, I set up followme on Asterisk 1.4.8 something like this and it >> > worked great: >> > >> > >> > from my followme.conf: >> > number=>207&206&5554441212,28 >> > >> > >> > Problem is after moving this same config to my new 1.8 box the call >> > fails and goes to voice mail if either of the two sip extensions are >> > unreachable. >> > >> > >> > So, let me explain further... >> > >> > >> > If both SIP/207 and SIP/206 are up and running and accessible to >> > receive the call then all goes well, if one of them is down for some >> > reason then none of the 3 extensions ring and it just goes to voice >> > mail. This stinks because I lose all calls to voice mail if for >> > example my Internet connection goes down at home (207). Wouldn't >> > this be the time you really want your other phones to ring? >> > >> > >> > I thought about doing something like this: >> > number=>207&206&5554441212,28 >> > number=>207&5554441212,28 >> > >> > >> > In case say 206 fails but when 206 is up, they will be on hold for >> > almost a minute before going to voice mail if I don't answer. >> > >> > >> > I know there are other solutions outside followme but this worked in >> > 1.4.8 and I have to think it should work in 1.8. Just not sure what >> > I am missing. >> > >> > >> > Thanks in advance for any help. >> >> This may work now after I fixed this issue last week on SVN v1.8: >> https://issues.asterisk.org/jira/browse/ASTERISK-17557 >> >> Do you get "Extension '%s@%s' doesn't exist\n" error messages? >> >> Richard >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
