(begin quote)
Installing from the Subversion repo (http://docs.yate.ro/wiki/Compiling_and_installing_Yate_from_SVN) is the best way to update. Later, when the programmers update the code, all you have to do is go back into the source directory, type "svn up" then make clean ; make ; make install-noapi.

Installing a new version won’t overwrite your config files. I believe there was a change to the ysipchan.conf if you customized the listen port. It used to go in the [listener general] section and now it just goes in the [general] section.
(end quote)

I took Bill’s advice and only had a couple of issues. The first was that when I tried to restart Yate it exited with an error:

./yate error while loading shared libraries: libyate.so.4.3.1: cannot open shared object file: No such file or directory

The fix for that was to do this:

echo "/usr/local/lib" > /etc/ld.so.conf.d/local.conf
ldconfig

Then I found that when one of my users placed a call using a softphone he did not hear ringback tone. His softphone displayed a message saying "early media" but that was all. Bill Simon also had a solution for that:

(begin quote)
On my Yate 4.3.1 installation, I have the following in regexroute.conf to get rid of the early media offer and send straight SIP 180s for ringing:

in the [extra] section, add: call.ringing=10

below the [extra] section, add a new section:

[call.ringing]
.*=;earlymedia=no

Then reload (connect to rmanager with "telnet localhost 5038″ or restart yate) and you should have ringing.
(end quote)

So with Bill’s help (thank you, Bill!) I was able to upgrade Yate and keep everything working. I wanted to pass this information along, since that thread is no longer accessible.

[WM: http://nerd.bz/VNMquK ]

callAccept [0x988a518]
disconnected. final=0 reason=noauth [0x988a518]
Hangup. reason=noauth [0x988a518]

Asterisk is registered to Yate with no problems, Yate is registered with Google no problems (outgoing calls work). Obviously Yate thinks that Asterisk is not authorized, but I can’t figure out where I missed a step….

dtmf_delay=2

and change the 2 to a 1, or you can even try 0, which seems to be working for me. Save the change and then do

service yate restart

to get it to re-read the file. Probably best to do it at a time when you know that no one will be in the middle of a call.

GV8431234567:TOPsecret@127.0.0.1:5050/8431234567

After registering to YATE with your client, dial the calls with 10-digits instead of 11. Remember you can’t be registered with Asterisk and a SIP client to the same Google Voice account at the same time on YATE (or any SIP server).

^\([0-9]\+\)$=jingle/1\1@voice.google.com;line=GV9991234567;ojingle_version=0;ojingle_flags=noping; redirectcount=5;checkcalled=false;dtmfmethod=rfc2833

Username: GV123456789
Password: (secure password used during setup)
IP: 192.168.1.100 (or whatever)

Does yate do this or do I need Asterisk for this?

Thanks,

Brian