Olle E. Johansson
2004-May-09 00:59 UTC
[Asterisk-Users] *** Asterisk sunday news: Read the sample configs, Luke!
* Read the config sample files! (even if you're an Asterisk guru) ----------------------------------------------------------------- For those of you that have a working installation that you keep using, this is a reminder to check into the configs/ directory of the Asterisk source tree, regardless if you downloaded a tar ball or from CVS. As we add or change features in Asterisk, the sample config files are updated. If you look there, you might get new insights into how to solve your problems. Also, you might find new features that you really need. If you have a new installation "gmake samples" or "make samples" will install these files for you. In CVS head, the development source tree, we've added quite a lot of information recently to these files. They are more educational now and contains a lot of sample configurations. *** Check app_groupcount! ------------------------- There's a new app in Asterisk town. In fact, there are several new applications in CVS head. One of the major recent additions is app_groupcount, that you can use to limit the number of calls to, well, just about anything. A SIP peer, a PRI link, a call center staff member, a conference and calls to or from your boy and/or girlfriend :-) The command for setting a group is setgroup(), the command for enforcing a limitation is checkgroup(). Please start using this instead of the incominglimit and outgoinglimit settings in sip.conf. These are not working as expected and the more general solution with app_groupcount is a much better solution that works cross channels. This is an end-of-life warning for outgoinglimit and incominglimit :-) As always, the CLI command "show applications" and "show application <name>" is your best friend. *** Set your SIP realm! ----------------------- In CVS head, the SIP channel is now able to use a proper SIP realm for authentication. The realm is the server group that has a common authentication for a user. It could be one server or a number of servers that shares a password/user database. According to the SIP RFC, it should be set either to a domain or a hostname, depending on what your realm covers. It should be globally unique. Up to know, all Asterisk servers used the "asterisk" realm. That made it a bit hard for some phones to know the difference between one server and another. Please note that if you are using the "md5secret" setting in sip.conf, this secret is based on the realm. If you change the realm, you need to rehash your secrets. *** Asterisk 1.0: Less than five bugs away ------------------------------------------ If you follow the CVS, you will notice that there are very few changes in the stable part of the source tree. Only bug fixes go in there and Mark have been working like crazy to fix the major bugs. The bug tracker had almost 300 open bugs just a while ago, and we are now down to a handful identified bugs. As usual with Open Source Software, relase is not set to marketing plans. Release will come when the software is ready to be shipped. So when Mark decides that we've fixed the bugs that needs fixing, a release candidate will be made and published for download. Please plan to help us test the 1.0rc1 real hard. Do whatever you can to crash it, to make it dial your mother-in-law when you really want to talk to your husband, to make it connect the whole office to the HR departments secret conference call by mistake and accidentally fill your hard disk drive with non-existing voice mail messages. We do not belive that you can, but if you can, report the bugs and help us move forward to a 1.0 release! If you want to start stress-testing it now, download the stable CVS release. Instructions is to be found at http://www.asterisk.org *** Astricon: Coming right up, sir! ----------------------------------- We get a lot of questions about Astricon. To answer a few: - We're still open of speaker's proposals, even though the time limit is up. - The conference venue is not set yet, we will add it to the web site as soon as we have more information - Pre-registration will start rsn (real soon now) - We will find a location with a standard class hotel as well as a lower price alternative. - Yes, we will have the voice of Asterisk there (hint, hint) Astricon is at http://www.astricon.net * Useful pointers: ------------------ * Asterisk: http://www.asterisk.org * Asterisk mailing lists: http://lists.digium.com (users, dev, biz and cvs mailing list) * Asterisk bug tracker: http://bugs.digium.com * Asterisk IRC channel: #asterisk on irc.freenode.net * Digium: http://www.digium.com * Wiki: http://www.voip-info.org * Voip Search: http://search.voip-forum.com * Astricon: http://www.astricon.net Have a nice Asterisk week! /Olle
Mark Elkins
2004-May-09 05:33 UTC
[Asterisk-Users] *** Asterisk sunday news: Read the sample configs, Luke!
On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote:> * Read the config sample files! (even if you're an Asterisk guru) > ----------------------------------------------------------------- > For those of you that have a working installation that you keep using, this is a > reminder to check into the configs/ directory of the Asterisk source tree, regardless > if you downloaded a tar ball or from CVS.Good advice - so I do a CVS UPDATE... and 'say.c' is broken.... gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/05/04-09:58:21\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -c -o say.o say.c say.c: In function `ast_say_digit_str': say.c:50: syntax error before '<<' token say.c:57: warning: no return statement in function returning non-void say.c: At top level: say.c:58: syntax error before "if" and in 'say.c' at about line 50.... case ('#'): snprintf(fn, sizeof(fn), "/digits/pound"); break; default: <<<<<<< say.c snprintf(fn, sizeof(fn), "/digits/%c", fn2[num]); } ====== if((fn2[num] >= '0') && (fn2[num] <'9')){ /* Must be in {0-9} */ snprintf(fn, sizeof(fn), "digits/%c", fn2[num]); } ------------------------------------------ The lines that begin with "<<<<<<<<<<< say.c" -or is this just an error caused by CVS ???? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ mje@posix.co.za - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040509/06338813/attachment.pgp