Re: [Asterisk-Dev] is this a bug?
Umm... -c is not for color... It is for console connection. Please check your documentation. - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com Sent: Tuesday, January 25, 2005 12:41 PM Subject: Re: [Asterisk-Dev] is this a bug? On January 25, 2005 12:59 pm, Steven Critchfield wrote: Ctrl-c is actually an interupt. Windows users needed easy to remember shortcuts, thats why you are used to ctrl-c. Yes but why does ^C only interrupt asterisk when -c (colour) is given? To me this is a bug. Asterisk should either always trap these key sequences or always pass them, irrespective of your desire to see coloured console messages. -A. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Memory Leak in Asterisk 1.0.0?
You may want to look into the Makefile, and compile in the ast_mm module to monitor that stuff. - Original Message - From: joachim [EMAIL PROTECTED] To: Asterisk Developers Mailing List [EMAIL PROTECTED] Sent: Sunday, December 05, 2004 2:34 PM Subject: Re: [Asterisk-Dev] Memory Leak in Asterisk 1.0.0? I dont see any memory leaks in my app_queue setup around 20.000 calls a day, so anything unless its a very small leak, there is no leak. (at least not in the parts i'm using.) Zoa. Steven Critchfield wrote: On Sat, 2004-12-04 at 10:14 -0500, Francois Lambert wrote: Does anyone have experience any sign of memory leak with Asterisk 1.0.0. I do use chan_agent (AgentLogin) and app_queue along with chan_iax2 quite intensively and I always finish the day with 125M of memory being used. As usual for this question, how are you measuring the memory usage? My main asterisk install has had asterisk running over 17 weeks yet memory usage for the whole of the machine is under 256 megs. We don't use any agents, just zap and iax channels, and AGI. That memory usage also includes a database that gets only a few queries a week. Granted I am pre 1.0, but I love to point out how stable pre 1.0 releases where. ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: Re[2]: [Asterisk-Dev] Added Postgresql support for SIP friends
Thanks for the support, Gunnar! Now I can add yet another tick to the count of people I know of using ast_data in their daily lives :-) I haven't had the time yet, to fully read-up on the app_realtime() etc. functions to see how they work. There are some things still to come in ast_data, but I have not had a lot of time lately. However, ast_data already has support for IAX2, SIP, Voicemail users, and Voicemail Directory. Rob - Original Message - From: Gunnar Schaller [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 06, 2004 3:44 PM Subject: Re[2]: [Asterisk-Dev] Added Postgresql support for SIP friends Right way. From the beginning of my asterisk server I use ast_data. All other is just to much scattered. A little bit here, a little bit there ... Gunnar Nice, but you may notice that all of the mysql code just got removed from chan_sip.c in the last 24 hours. :-) - Original Message - From: Jens Kübler [EMAIL PROTECTED] To: Asterisk Developers Mailing List [EMAIL PROTECTED] Sent: Wednesday, October 06, 2004 12:26 PM Subject: [Asterisk-Dev] Added Postgresql support for SIP friends It's untested but should work out of the box as it is a straight copy of the mysql stuff. I added some transaction stuff as someone else might change sipfriends during reading the db. For me it compiled fine. I will appreciate, if someone looks into the Makefile stuff as I'm not pretty familiar with that. I've added support for linux only. Be sure to add pgconnection=. in sip.conf to connect. Jens ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Added Postgresql support for SIP friends
See another reply I just posted, James. ast_data is modularized in order to do that. app_realtime() could be. I haven't read it yet. But, ast_data supports mysql, pgsql, and odbc out of the box. You can also mix and match them if you wanted (not that you would want to) Rob - Original Message - From: James Sharp [EMAIL PROTECTED] To: Asterisk Developers Mailing List [EMAIL PROTECTED] Sent: Wednesday, October 06, 2004 12:49 PM Subject: Re: [Asterisk-Dev] Added Postgresql support for SIP friends Shouldn't it be migrated to ODBC as well? Database abstraction is a good thing. On Oct 6, 2004, at 1:26 PM, Jens Kübler wrote: It's untested but should work out of the box as it is a straight copy of the mysql stuff. I added some transaction stuff as someone else might change sipfriends during reading the db. For me it compiled fine. ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Good Explain about Virus in Voip.
Of course, a computer CAN be infected by a WORM through VoIP If someone programs a VoIP client that contains a buffer overflow problem (read: Microsoft written software), then anyone could write a program to exploit the overflow, and voila... worm ANY kind of network client/server software is vulnerable to attacks of this type. This is nothing new just because it is VoIP. The way you protect against this is through good network programming, and paying careful attention to any functions that copy memory around (IE: memcpy(), strcpy(), sprintf(), etc...) - Original Message - From: Carlos Arnt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 1:21 PM Subject: [Asterisk-Dev] Good Explain about Virus in Voip. Did someone has a good explanation about why Voip can't be infected by virus ?? Ok Maybe using only hardware stuff and has * that is a linux is good enough but for the Windows clients programs ?? Someone can explain than for me why then we can be just worry about this ? Thanks alot ! Oh yes whats happend with asterisk-users list ??? Long time i can't receive nothing. Thanks again. ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] ast_data pgsql
:-) I see the error in the code now. I am fixing it, and maybe by the time you get this, it will be available at http://svn.asteriskdocs.org/res_data The issue is that the Default SQL I changed code to a little while back has a SELECT *, but the code to read the fields is positional for that data_pgsql function Eventually, it will scan the row for the field name it needs, and this will make it more able to handle queries of this sort. Rob - Original Message - From: Dmitri Pavlenkov [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 16, 2004 2:57 PM Subject: [Asterisk-Dev] ast_data pgsql I think everything's setup correctly, and yet extension lookup is not working: iptel1= \d extensions Table public.extensions Column| Type | Modifiers -++- -- context | character varying(40) | not null default 'default'::character varying extension | character varying(20) | not null priority| integer| not null default 1 application | character varying(20) | not null args| character varying(100) | not null default ''::character varying Indexes: extensions_pkey primary key btree (context, extension, priority) iptel1= select * from extensions iptel1- ; context | extension | priority | application | args -+---+--+-+- dbtest | s |1 | Festival| test successful im1 | 999 |1 | Goto| dbtest|s|1 (2 rows) paska*CLI show data pgsql status == Database: [EMAIL PROTECTED]:5432 Queries Handled: EXTENSION_LOOKUP ID In/Out Reads Writes Deletes Status 1 In 8 0 0 Connected 0 days, 00:03:48 2 In 0 0 0 Connected 0 days, 00:03:48 3 In 0 0 0 Connected 0 days, 00:03:48 == == Request: == Packet Dump == Query Type: SIP_FIND_USER == (100)Name = (STR)dmitri == End of Packet Dump == Response: == Request: == Packet Dump == Query Type: SIP_FIND_USER == (100)Name = (STR)dmitri == End of Packet Dump == Response: == Request: == Packet Dump == Query Type: SIP_FIND_USER == (100)Name = (STR)dmitri == End of Packet Dump == Response: == Request: == Packet Dump == Query Type: EXTENSION_LOOKUP == (114)Extension = (STR)999 == (110)Context = (STR)im1 == (115)Priority = (INT)1 == (134)Action = (INT)0 == End of Packet Dump == Response: == Packet Dump == Query Type: EXTENSION_LOOKUP == (20)DataSource = (STR)pgsql == (10)ErrorNumber = (INT)-9 == (11)ErrorMessage = (STR)Not found == End of Packet Dump == Request: == Packet Dump == Query Type: EXTENSION_LOOKUP == (114)Extension = (STR)999 == (110)Context = (STR)im1 == (115)Priority = (INT)1 == (134)Action = (INT)3 == End of Packet Dump == Response: == Packet Dump == Query Type: EXTENSION_LOOKUP == (20)DataSource = (STR)pgsql == (114)Extension = (STR)im1 == (116)Application = (STR)999 == (117)Arguments = (STR)1 == (10)ErrorNumber = (INT)0 == (11)ErrorMessage = (STR)OK == End of Packet Dump == Request: == Packet Dump == Query Type: SIP_FIND_USER == (100)Name = (STR)dmitri == End of Packet Dump == Response: Dmitri Pavlenkov Technical Support Im1 Web Hosting ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] ast_data pgsql
In fact, I committed one piece of code (that would work) by changing the SQL. Then I decided to just fix it permanently. The SQL is back to SELECT *, but the code now checks the field names instead of just their positions. This change is going to be needed anyhow, when I finally complete the configurable SQL portion of the program. Rob PS: Thanks for finding this Dimitri - Original Message - From: Rob Gagnon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 16, 2004 4:33 PM Subject: Re: [Asterisk-Dev] ast_data pgsql :-) I see the error in the code now. I am fixing it, and maybe by the time you get this, it will be available at http://svn.asteriskdocs.org/res_data The issue is that the Default SQL I changed code to a little while back has a SELECT *, but the code to read the fields is positional for that data_pgsql function Eventually, it will scan the row for the field name it needs, and this will make it more able to handle queries of this sort. Rob - Original Message - From: Dmitri Pavlenkov [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 16, 2004 2:57 PM Subject: [Asterisk-Dev] ast_data pgsql I think everything's setup correctly, and yet extension lookup is not working: iptel1= \d extensions Table public.extensions Column| Type | Modifiers -++- -- context | character varying(40) | not null default 'default'::character varying extension | character varying(20) | not null priority| integer| not null default 1 application | character varying(20) | not null args| character varying(100) | not null default ''::character varying Indexes: extensions_pkey primary key btree (context, extension, priority) iptel1= select * from extensions iptel1- ; context | extension | priority | application | args -+---+--+-+- dbtest | s |1 | Festival| test successful im1 | 999 |1 | Goto| dbtest|s|1 (2 rows) paska*CLI show data pgsql status == Database: [EMAIL PROTECTED]:5432 Queries Handled: EXTENSION_LOOKUP ID In/Out Reads Writes Deletes Status 1 In 8 0 0 Connected 0 days, 00:03:48 2 In 0 0 0 Connected 0 days, 00:03:48 3 In 0 0 0 Connected 0 days, 00:03:48 == == Request: == Packet Dump == Query Type: SIP_FIND_USER == (100)Name = (STR)dmitri == End of Packet Dump == Response: == Request: == Packet Dump == Query Type: SIP_FIND_USER == (100)Name = (STR)dmitri == End of Packet Dump == Response: == Request: == Packet Dump == Query Type: SIP_FIND_USER == (100)Name = (STR)dmitri == End of Packet Dump == Response: == Request: == Packet Dump == Query Type: EXTENSION_LOOKUP == (114)Extension = (STR)999 == (110)Context = (STR)im1 == (115)Priority = (INT)1 == (134)Action = (INT)0 == End of Packet Dump == Response: == Packet Dump == Query Type: EXTENSION_LOOKUP == (20)DataSource = (STR)pgsql == (10)ErrorNumber = (INT)-9 == (11)ErrorMessage = (STR)Not found == End of Packet Dump == Request: == Packet Dump == Query Type: EXTENSION_LOOKUP == (114)Extension = (STR)999 == (110)Context = (STR)im1 == (115)Priority = (INT)1 == (134)Action = (INT)3 == End of Packet Dump == Response: == Packet Dump == Query Type: EXTENSION_LOOKUP == (20)DataSource = (STR)pgsql == (114)Extension = (STR)im1 == (116)Application = (STR)999 == (117)Arguments = (STR)1 == (10)ErrorNumber = (INT)0 == (11)ErrorMessage = (STR)OK == End of Packet Dump == Request: == Packet Dump == Query Type: SIP_FIND_USER == (100)Name = (STR)dmitri == End of Packet Dump == Response: Dmitri Pavlenkov Technical Support Im1 Web Hosting ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL
Re: [Asterisk-Dev] ast_data, mysql, md5secret
RE: checking for .conf file existence Good idea RE: username updating to name This is by design in the original chan_sip.c code, and is mirrored in ast_data. The username from the client is what is saved in the username field. Not sure why, but thats how it was in the past. Rob - Original Message - From: Gunnar Schaller [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 16, 2004 5:58 AM Subject: Re[2]: [Asterisk-Dev] ast_data, mysql, md5secret One more question: I saved a sip-user in the database and set name to and username to abc. When the client logs in asterisk makes an update to the database and sets username to . Why that? Any reasons? Gunnar ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] ast_data, mysql, md5secret
Oddly enough, some code was in the modules for md5secret support, but I had commented it out a while back because it was not made optional at the time. It is now supported in all data_xxx.c modules as well as in the chan_sip.c.patch.txt file. Thanks for the hint Gunnar! Rob - Original Message - From: Gunnar Schaller [EMAIL PROTECTED] To: Gunnar Schaller [EMAIL PROTECTED] Sent: Tuesday, June 15, 2004 1:45 PM Subject: Re: [Asterisk-Dev] ast_data, mysql, md5secret Ups, just wanted to say everything works fine ... I am running asterisk cvs version patched with newest ast_data. My database is in MySQL, and there is my problem. Anything works fine, -- ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Centralized voicemail
Soren, Information about the methods followed to get to the error, as well as any log messages prior to it would be nice. Also, if you know how to use GDB, that would be helpful to get the last few things that happened. Rob - Original Message - From: Soren Rathje [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 11, 2004 12:17 PM Subject: Re: [Asterisk-Dev] Centralized voicemail - Original Message - From: brian k. west [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 4:59 AM Subject: Re: [Asterisk-Dev] Centralized voicemail its ast_data now.. its been brought into the core of asterisk. bkw Brian, I get a segfault on VoiceMail with ast_data version 133, 134, 136, 141 with current HEAD 11-06-04. Plain vanilla FC1, MySQL, PostgreSQL, unixODBC. segfault reproduceable every time on MySQL, PgSQL and ODBC(PgSQL). What do you need to debug, when it comes to Linux I'm still a virgin.. :-) -- Soren ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] app_meetme crash
Here's the problem
Line 928 of v1.36 app_meetme.c:
cur = confs;
if (!conf-users) {
...
} else {
/* Remove the user struct */
if (user == cur-firstuser) {
..
The issue is that the Remove the user struct section thinks that cur- is
same as conf- at this point, but it is actually confs (with an s)
The entire else block should change cur to conf to become:
/* Remove the user struct */
if (user == conf-firstuser) {
conf-firstuser-nextuser-prevuser = NULL;
conf-firstuser = conf-firstuser-nextuser;
} else if (user == conf-lastuser){
conf-lastuser-prevuser-nextuser = NULL;
conf-lastuser = conf-lastuser-prevuser;
} else {
user-nextuser-prevuser = user-prevuser;
user-prevuser-nextuser = user-nextuser;
}
So that the user is removed from the current conference, and not from
confs which is the head.
- Original Message -
From: Rob Gagnon [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 02, 2004 5:08 PM
Subject: Re: [Asterisk-Dev] app_meetme crash
Hmmm
If user-prevuser is NULL, then it SHOULD be the head of the list, in
which
case the if (user == cur-firstuser) statement SHOULD have kicked in.
So... this leads me to believe there is another bug some place else
actually
causing the problem.
I am not that familiar with the inerds of app_meetme, but from what I know
of doubly-linked-lists, the if-statement you added should not be needed if
the list integrity is maintained.
Rob
- Original Message -
From: Jared Mauch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 02, 2004 4:29 PM
Subject: [Asterisk-Dev] app_meetme crash
See the patch, this will prevent it from coring, but could lead
to other issues.
- Jared
#0 0x003d2c1e in conf_run (chan=0x88aa400, conf=0x86ee8d8,
confflags=536)
at app_meetme.c:942
942 user-prevuser-nextuser =
user-nextuser;
(gdb) print *user
$1 = {user_no = 1, prevuser = 0x0, nextuser = 0x88beba8, userflags =
536,
adminflags = 0, chan = 0x88aa400,
usrvalue = test, '\0' repeats 45 times, jointime = 2586960}
(gdb) print *user-prevuser
Cannot access memory at address 0x0
(gdb) print *user-nextuser
$2 = {user_no = 2, prevuser = 0x0, nextuser = 0x8694058, userflags =
536,
adminflags = 0, chan = 0x8878370,
usrvalue = test, '\0' repeats 45 times, jointime = 2586960}
(gdb) print user-nextuser-prevuser
$3 = (struct ast_conf_user *) 0x0
(gdb) print user-prevuser
$4 = (struct ast_conf_user *) 0x0
(gdb) print user-nextuser
$5 = (struct ast_conf_user *) 0x88beba8
(gdb) print user-prevuser
$6 = (struct ast_conf_user *) 0x0
diff -u -r1.35 app_meetme.c
--- app_meetme.c1 Jun 2004 22:54:18 - 1.35
+++ app_meetme.c2 Jun 2004 21:27:24 -
@@ -939,7 +939,9 @@
cur-lastuser = cur-lastuser-prevuser;
} else {
user-nextuser-prevuser =
user-prevuser;
- user-prevuser-nextuser =
user-nextuser;
+ if (user-prevuser != NULL) {
+ user-prevuser-nextuser =
user-nextuser;
+ }
}
/* Return the number of seconds the user was in
the conf */
sprintf(meetmesecs, %i, (int)
(user-jointime -
time(NULL)));
--
Jared Mauch | pgp key available via finger from [EMAIL PROTECTED]
clue++; | http://puck.nether.net/~jared/ My statements are only
mine.
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Re: [Asterisk-Dev] app_meetme crash
Here's a patch to fix the logical bug:
Is there something open on bugs.digium.com??
(sorry for the small amount of re-format changes, but I had to align the { }
sets to see the logic better)
Rob
Index: app_meetme.c
===
RCS file: /usr/cvsroot/asterisk/apps/app_meetme.c,v
retrieving revision 1.36
diff -u -p -r1.36 app_meetme.c
--- app_meetme.c2 Jun 2004 16:57:14 - 1.36
+++ app_meetme.c2 Jun 2004 22:57:50 -
@@ -918,7 +918,7 @@ zapretry:
outrun:
if (user-user_no) { /* Only cleanup users who really joined! */
- manager_event(EVENT_FLAG_CALL, MeetmeLeave,
+ manager_event(EVENT_FLAG_CALL, MeetmeLeave,
Channel: %s\r\n
Uniqueid: %s\r\n
Meetme: %s\r\n,
@@ -926,34 +926,34 @@ outrun:
ast_mutex_lock(conflock);
conf-users--;
cur = confs;
- if (!conf-users) {
- /* No more users -- close this one out */
- while(cur) {
- if (cur == conf) {
- if (prev)
- prev-next = conf-next;
- else
- confs = conf-next;
- break;
+ if (!conf-users) {
+ /* No more users -- close this one out */
+ while(cur) {
+ if (cur == conf) {
+ if (prev)
+ prev-next = conf-next;
+ else
+ confs = conf-next;
+ break;
+ }
+ prev = cur;
+ cur = cur-next;
}
- prev = cur;
- cur = cur-next;
- }
- if (!cur)
- ast_log(LOG_WARNING, Conference not found\n);
- if (conf-chan)
- ast_hangup(conf-chan);
- else
- close(conf-fd);
- free(conf);
+ if (!cur)
+ ast_log(LOG_WARNING, Conference not
found\n);
+ if (conf-chan)
+ ast_hangup(conf-chan);
+ else
+ close(conf-fd);
+ free(conf);
} else {
/* Remove the user struct */
- if (user == cur-firstuser) {
- cur-firstuser-nextuser-prevuser = NULL;
- cur-firstuser = cur-firstuser-nextuser;
- } else if (user == cur-lastuser){
- cur-lastuser-prevuser-nextuser = NULL;
- cur-lastuser = cur-lastuser-prevuser;
+ if (user == conf-firstuser) {
+ user-nextuser-prevuser = NULL;
+ conf-firstuser = user-nextuser;
+ } else if (user == conf-lastuser){
+ user-prevuser-nextuser = NULL;
+ conf-lastuser = user-prevuser;
} else {
user-nextuser-prevuser = user-prevuser;
user-prevuser-nextuser = user-nextuser;
- Original Message -
From: Fabian Stelzer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 02, 2004 5:24 PM
Subject: Re: [Asterisk-Dev] app_meetme crash
Yeah the if is normally not needed!
can you provide more information about the situation of the crash? (how
many
users where in, who left or anything else (your usage of meetme)).
perhaps i can reproduce this and then correctly fix it! (i made the patch)
Regards
Fabe
- Original Message -
From: Rob Gagnon [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 03, 2004 12:08 AM
Subject: Re: [Asterisk-Dev] app_meetme crash
Hmmm
If user-prevuser is NULL, then it SHOULD be the head of the list, in
which
case the if (user == cur-firstuser) statement SHOULD have kicked in.
So... this leads me to believe there is another bug some place else
actually
causing the problem.
I am not that familiar with the inerds of app_meetme, but from what I
know
of doubly-linked-lists, the if-statement you added should not be needed
if
the list integrity is maintained.
Rob
- Original Message -
From: Jared Mauch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 02, 2004 4:29 PM
Subject: [Asterisk-Dev] app_meetme crash
See the patch
Re: [Asterisk-Dev] Security Issue in Asterisk with sip.conf configuration.
Have you tried using: permit= deny= entries in the sip.conf file? you can have as many of those as you need to create an ACL - Original Message - From: William Zhang [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, April 27, 2004 5:31 PM Subject: [Asterisk-Dev] Security Issue in Asterisk with sip.conf configuration. I had tried many ways with some advanced user help, but without success(at one point I thought I had it worked). Here Asterisk is working as a SIP PSTN Gateway, and in the sip.conf file, there are a lot of entries with just host=a.b.c.d, thinking that * will only accept calls from host a.b.c.d, but in my test, no mater how you set up the sip.conf entries, either * will NOT accept calls for that user account at all, or it will accept calls from any where without VERIFYING the source IP(whether it is a.b.c.d or not), so long the sip userid is the username in sip.conf. This post a very serious security problem. Of course we can put secret= for each entries, but giving Asterisk GW and SIP proxy are in 2 TRUSTED IPs, no Authentication is neccessary, otherwise it increase the SIP traffic quite a bit. Following are the 4 different entries that I had tried: #Notice that in the general section, context is pointed to a none existant context INVALID. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 212.213.66.68 context = INVALID ; ;srvlookup = yes; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=g729 allow=ilbc ; ;dtmfmode=info ;dtmfmode=inband dtmfmode=rfc2833 [20034] type=friend callerid=TEST 61331045 host=212.213.65.66 nat=yes; This phone may be natted canreinvite=no [20035] type=peers callerid=TEST 61331045 host=212.213.65.66 nat=yes; This phone may be natted canreinvite=no [20036] type=friend context=default callerid=TEST 61331045 host=212.213.65.66 permit=212.213.65.66 nat=yes; This phone may be natted canreinvite=no [20037] type=peers context=default callerid=TEST 61331045 permit=212.213.65.66 nat=yes; This phone may be natted canreinvite=no Thank you in advance. ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Makefile:73: .depend: No such file or directory
I'm not an expert on the compilation environment you need, but if you provide the OS you are under (don't say Linux) and the versions of the gcc stuff you have, it might help someone else on this list help you. It appears to me to be a missing environment variable, or compiler compatibility problem. Also, have you made sure the following dependencies are met? rpm -q kernel-source readline readline-devel openssl openssl-devel - Original Message - From: karunesha [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 21, 2004 5:33 AM Subject: Re: [Asterisk-Dev] Makefile:73: .depend: No such file or directory Hi Rob Garnon I have rewritten the question, Actualy i had downloded Asterisk DEVELOPER Version from the CVS with the help of SysAdmin. Downloded Zaptel. libpri and asterisk.And placed all the folders in our linux system /usr/src/ directory. 1. I tried to install the Zaptel and did it successfuly. But at the end of installation i seen following lines. What does it meen, have i missed any files or some thing else. /sbin/depmod -a depmod: *** Unresolved symbols in /lib/modules/2.4.2-2/kernel/drivers/char/ocdemonpp.o depmod: *** Unresolved symbols in /lib/modules/2.4.2-2/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.2-2/misc/zaptel.o [ -f /etc/zaptel.conf ] || install -m 644 zaptel.conf.sample /etc/zaptel.conf 2. Next i tried to install the libpri, it is displaying an error message 'Makefile:73: .depend: No such file or directory'. Following is the log data. Can you please tell me why i am getting this error. I searched .depend file but i didn't find anywhere. [EMAIL PROTECTED] libpri]# make clean Makefile:73: .depend: No such file or directory ./mkdep -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g `ls *.c` : unrecognized option Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ... GNU long options: --debug --dump-po-strings --dump-strings --help --login --noediting --noprofile --norc --posix --rcfile --rpm-requires --restricted --verbose --version --wordexp Shell options: -irsD or -c command (invocation only) -abefhkmnptuvxBCHP or -o option make: *** [.depend] Error 2 3. Next i tried to clean and install asterisk, i am getting the following log telling Error. [EMAIL PROTECTED] asterisk]# make clean for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x clean || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/res' make[1]: Entering directory `/usr/src/asterisk/channels' rm -f *.so *.o .depend rm -f busy.h ringtone.h gentone gentone-ulaw make[1]: Leaving directory `/usr/src/asterisk/channels' @make[1]: Entering directory `/usr/src/asterisk/pbx' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/pbx' make[1]: Entering directory `/usr/src/asterisk/apps' rm -f *.so *.o look .depend make[1]: Leaving directory `/usr/src/asterisk/apps' make[1]: Entering directory `/usr/src/asterisk/codecs' Makefile:89: .depend: No such file or directory ../mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-dec larations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march= i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/21/04-15:36:55\ -D INSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/ast erisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DA STSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCO NFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modul -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN - DNEW_PRI_HANGUP -fPIC `ls *.c` : unrecognized option Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ... GNU long options: --debug --dump-po-strings --dump-strings --help --login --noediting --noprofile --norc --posix --rcfile --rpm-requires --restricted --verbose --version --wordexp Shell options: -irsD or -c command (invocation only) -abefhkmnptuvxBCHP or -o option make[1]: *** [.depend] Error 2 make[1]: Leaving directory `/usr/src/asterisk/codecs' make: *** [clean] Error 1 [EMAIL PROTECTED] asterisk]# make install ./mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-decl arations -g -Ii clude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OP TIMIZATIONS -DAS TERISK_VERSION=\CVS-04/21/04-15:34:38\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\
Re: [Asterisk-Dev] Grandstream TFTP configuration.
Are the cfgmac file contents correct? They normally require a binary encrypted file, not just a text file with values. At least thats how the ATA18x's work. I'd imagine it is similar for GS - Original Message - From: chaye wala [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, April 02, 2004 12:58 AM Subject: [Asterisk-Dev] Grandstream TFTP configuration. I read all the messages on the Mail Thread but still am unable to configure the GS BT-100 with a TFTP server. GS loads the .54 version of boot files but never takes any configuration changes. I see it is loading cfgmac file but ignores the contents. Any tips? What is needed to make GS BT100 configure using a TFTP server. Thanks. __ Do you Yahoo!? Yahoo! Small Business $15K Web Design Giveaway http://promotions.yahoo.com/design_giveaway/ ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] New Query API I wrote... want to discuss inclusion:
I hope I didn't spam people with this. I sent the original from the wrong email address: After debating with citats for a bit in IRC, his idea of an abstract query interface finally made sense to me :-) So I set about writing it on the weekend, and it is in a working state now. With this API (which I named "qint" for query interface), (similar to how cdr.c works) 1) All mySQL, and other DB specificcode can be removed from being all over Asterisk (voicemail, chan_sip, directory, etc.) 2) All the #ifdef MYSQL lines of code can be removed from Asterisk 3) Modules (ie: qint/qint_mysql.c, or qint_odbc.c, or qint_radius.c) can be written that contain the specific code for the lookups that are moved to qint. 4) A Single database connection can be shared by qint, instead of having one for each piece that connects 5) Configuring the database connection doesnt have to be in 3 or 4 different .conf files So, here's a quick list of the changes... with more discussion, we can figure out where to put code for others to review: - new file qint.c compiles into Asterisk just like cdr.c does, with some of its functions exported in qint.h - new file qint_mysql.c which can handle any query Asterisk would normally do from itself. (example: I already removed ALL mysql code from chan_sip.c and put it int qint_mysql.c best part is it still all works as if the code were right inside of chan_sip.c) - changed chan_sip.c and removed all mysql code, and #ifdefs. replaced with code like: //search for user/peer code as it exists ... if (!u) qint_sip_find_user(u, other params...); ... the qint_sip_find_user() function exists within qint.c:It will dynamically find a possible registered function for performing a sipfriends lookup in an external .so module (ie: qint_mysql.so) qint_mysql.so's load_module() function just performs a qint_register(name, desc, query_type, function_ptr) call in order to register a function as a handler for a specific type of query. Of course, this did require some other minor changes - Had to make a chan_sip.h file to define some things external of chan_sip.c (so both chan_sip.c, and qint_mysql.c can see the sip_peer and sip_user structs for example) - a #include asterisk/qint.h needed to be added to chan_sip.c and so on... lots of little things. This helps moving toward a dynamic db lookup for the extensions.conf file as well. With this, I was able to catch any query for pbx_find_extension(), which was a 2 line change to pbx.c, and a handler in qint_mysql.c to catch the call. If anyone would like to review the source I have, just reply. #asterisk - rgagnon
