Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
Really, I think we're pretty positive there's a ref leak (since
otherwise, the CBAnn channel would be long gone). If you can get a
ref debug log and the standard Asterisk DEBUG log showing the
problem, that would help a lot in finding out what is going on.
I think the bug is in conf_handle_talker_cb. It calls ao2_find but has no
mechanism to decremement the refcount. It appears that the following is
the best fix. I looked at all remaining calls to ao2_find in app_confbridge.c
and they look OK. Do you agree with the below fix?
*** app_confbridge.c.bug2014-05-06 06:42:21.0 -0400
--- app_confbridge.c2014-05-06 06:42:05.0 -0400
*** static int conf_handle_talker_cb(struct
*** 1461,1467
struct pvt_talker_cb *pvt = hook_pvt;
const char *conf_name = pvt-conf_name;
! struct confbridge_conference *conference = ao2_find(conference_bridges,
conf_name, OBJ_KEY);
struct ast_json *talking_extras;
if (!conference) {
/* Remove the hook since the conference does not exist. */
--- 1461,1468
struct pvt_talker_cb *pvt = hook_pvt;
const char *conf_name = pvt-conf_name;
! RAII_VAR(struct confbridge_conference *, conference, NULL, ao2_cleanup);
struct ast_json *talking_extras;
+ conference = ao2_find(conference_bridges, conf_name, OBJ_KEY);
if (!conference) {
/* Remove the hook since the conference does not exist. */
--
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Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
On Tue, May 6, 2014 at 5:45 AM, Richard Kenner ken...@gnat.com wrote:
Really, I think we're pretty positive there's a ref leak (since
otherwise, the CBAnn channel would be long gone). If you can get a
ref debug log and the standard Asterisk DEBUG log showing the
problem, that would help a lot in finding out what is going on.
I think the bug is in conf_handle_talker_cb. It calls ao2_find but has no
mechanism to decremement the refcount. It appears that the following is
the best fix. I looked at all remaining calls to ao2_find in
app_confbridge.c
and they look OK. Do you agree with the below fix?
*** app_confbridge.c.bug2014-05-06 06:42:21.0 -0400
--- app_confbridge.c2014-05-06 06:42:05.0 -0400
*** static int conf_handle_talker_cb(struct
*** 1461,1467
struct pvt_talker_cb *pvt = hook_pvt;
const char *conf_name = pvt-conf_name;
! struct confbridge_conference *conference =
ao2_find(conference_bridges, conf_name, OBJ_KEY);
struct ast_json *talking_extras;
if (!conference) {
/* Remove the hook since the conference does not exist. */
--- 1461,1468
struct pvt_talker_cb *pvt = hook_pvt;
const char *conf_name = pvt-conf_name;
! RAII_VAR(struct confbridge_conference *, conference, NULL,
ao2_cleanup);
struct ast_json *talking_extras;
+ conference = ao2_find(conference_bridges, conf_name, OBJ_KEY);
if (!conference) {
/* Remove the hook since the conference does not exist. */
--
That is definitely a leak and the fix looks good. That leak is most likely
the
one biting you. There is another leak in handle_cli_confbridge_kick() if
the
participant to kick is not in the conference.
Please go ahead and open an issue so proper credit can be given for the
patch.
Richard
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Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
That is definitely a leak and the fix looks good. Thanks. That leak is most likely the one biting you. It definitely is. There is another leak in handle_cli_confbridge_kick() if the participant to kick is not in the conference. Confirmed. I missed that one in my code reading. I just fixed it the same way. Please go ahead and open an issue so proper credit can be given for the patch. I'm not concerned about credit, but would like to get it fixed. I need to figure out what has to happen for me to be able to submit patches, but then I'll have some others to submit too. -- _ -- 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
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI ad...@tootai.net wrote: snip As explained in one on my previous message, it's a bug, easily reproducible: take a queues.conf (or sip.conf or iax.conf or voicemail.conf or ...) like this (what is important is the #include): snip NOTICE[3346]: app_queue.c:6811 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. despite the fact that modification was done in a .conf file. I took this example as with module reload app_queue the above message appears. For sip, iax, voicemail, aso there is no message, just SIP reload or ... To make asterisk take the modification in account, you have to open /etc/asterisk/[sip|iax|voicemail|queue|..].conf and save it without making any change. After this the command will be execute. It you run it a second time in a raw, you will see that the false behavior appears again till you again open/save the original file. Hi! I tried to reproduce using your description here and could not reproduce the issue. I tried with both sip.conf and queues.conf. Making a change in an included .conf file, but NOT the parent .conf file and then reloading that module from the CLI results in: centosclean*CLI module reload app_queue.so -- Reloading module 'app_queue.so' (True Call Queueing) [May 6 17:51:39] NOTICE[16211]: app_queue.c:7765 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. == Parsing '/etc/asterisk/queues.conf': Found == Parsing '/etc/asterisk/queue_include_1.conf': Found == Parsing '/tmp/queue_include_2.conf': Found I get the same behavior with sip.conf, it appears to work fine, whether I'm making only changes in the parent .conf or the included children. I even tried with two different included files in each sip.conf and queues.conf, one in /tmp and one in /etc/asterisk. Same working behavior. I used SVN-branch-11-r413305, so you might want to test there. However I'm still confused as to how you are seeing the behavior you are seeing. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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
Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI ad...@tootai.net wrote: snip As explained in one on my previous message, it's a bug, easily reproducible: take a queues.conf (or sip.conf or iax.conf or voicemail.conf or ...) like this (what is important is the #include): snip NOTICE[3346]: app_queue.c:6811 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. On Tue, 6 May 2014, Rusty Newton wrote: However I'm still confused as to how you are seeing the behavior you are seeing. Any chance the OP is including files from a file system that isn't maintaining atime/ctime/mtime/etc as expected, like NFS? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
[asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem
Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY server*CLI core show channel PJSIP/7000-0001 -- General -- Name: PJSIP/7000-0001 Type: PJSIP UniqueID: 1399382022.1 LinkedID: 1399382022.0 Caller ID: 7000 Caller ID Name: (N/A) Connected Line ID: 7001 Connected Line ID Name: 7001 Eff. Connected Line ID: 7001 Eff. Connected Line ID Name: 7001 DNID Digits: (N/A) Language: de State: Up (6) NativeFormats: (alaw) WriteFormat: g722 ReadFormat: g722 WriteTranscode: Yes (g722)-(slin)-(alaw) ReadTranscode: Yes (alaw)-(slin)-(g722) Time to Hangup: 0 Elapsed Time: 0h3m24s Bridge ID: 0cbdb7d7-81ed-4c5f-896f-cea37599b148 -- PBX -- Context: outgoing-kamailio Extension:pjsi Priority: 1 Call Group: 0 Pickup Group: 0 Application: AppDial Data: (Outgoing Line) Call Identifer: [C-] Variables: BRIDGEPEER=PJSIP/7001- DIALEDPEERNUMBER=7000 CDR Variables: level 1: calledsubaddr= level 1: callingsubaddr= level 1: dnid= level 1: clid= 700 level 1: src=7000 level 1: dcontext=outgoin level 1: channel=PJSIP/7 level 1: lastapp=AppDial level 1: lastdata=(Outgoi level 1: start=1399382 level 1: answer=1399382 level 1: end=1399382 level 1: duration=1 level 1: billsec=0 level 1: disposition=8 level 1: amaflags=3 level 1: uniqueid=1399382 level 1: linkedid=1399382 level 1: sequence=1 server*CLI core show channel PJSIP/7001- -- General -- Name: PJSIP/7001- Type: PJSIP UniqueID: 1399382022.0 LinkedID: 1399382022.0 Caller ID: 7001 Caller ID Name: 7001 Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: de State: Up (6) NativeFormats: (g722) WriteFormat: g722 ReadFormat: g722 WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h3m51s Bridge ID: 0cbdb7d7-81ed-4c5f-896f-cea37599b148 -- PBX -- Context: outgoing-kamailio Extension: 7000 Priority: 1 Call Group: 0 Pickup Group: 0 Application: Dial Data: PJSIP/7000 Call Identifer: [C-] Variables: BRIDGEPEER=PJSIP/7000-0001 DIALEDPEERNUMBER=7000 DIALEDPEERNAME=PJSIP/7000-0001 DIALSTATUS=ANSWER DIALEDTIME= ANSWEREDTIME= CDR Variables: level 1: calledsubaddr= level 1: callingsubaddr= level 1: dnid= level 1: clid=7001 level 1: src=7001 level 1: dst=7000 level 1: dcontext=outgoin level 1: channel=PJSIP/7 level 1: dstchannel=PJSIP/7 level 1: lastapp=Dial level 1: lastdata=PJSIP/7 level 1: start=1399382 level 1: answer=1399382 level 1: end=0.0 level 1: duration=230 level 1: billsec=228 level 1: disposition=8 level 1: amaflags=3 level 1: uniqueid=1399382 level 1: linkedid=1399382 level 1: sequence=0 -- _ -- 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
Re: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem
PS. if I configure both extension 7000 and 7001 to, disallow=all allow=alaw or disallow=all allow=g722 everything is fine. as long as the allowed codec is equal in both extensions. Am 07.05.2014 07:00, schrieb Rainer Piper: Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de -- _ -- 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
Re: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem
that's funny I recompiled asterisk without bridge_native_rtp.so to force asterisk to go to simple_bridge and not to native_bridge... !!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu Am 07.05.2014 07:11, schrieb Rainer Piper: PS. if I configure both extension 7000 and 7001 to, disallow=all allow=alaw or disallow=all allow=g722 everything is fine. as long as the allowed codec is equal in both extensions. Am 07.05.2014 07:00, schrieb Rainer Piper: Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de NOC +49 228 97167161 - sip.soho-piper.de NOC +882 990111550 via e164.org International Network NOC +49 2247 9064188 - sip.tefonix.de - D293 NOC +882 990045450 via e164.org International Network -- _ -- 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
Re: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem
perhaps a silly question ... if a channel switches from simple_bridge to native_bridge ... is the channel switching to direct_media between the endpoints ? if so, why doesn't turn direct_media = no and disable_direct_media_on_nat = yes switching to native_bridge off ? my pjsip.conf endpoint 7000 and 7001 [7000] type=endpoint context=outgoing disallow=all allow=alaw,ulaw,g722 transport=transport-udp auth=auth7000 aors=7000 direct_media = no disable_direct_media_on_nat = yes [auth7000] type=auth auth_type=userpass password=x username=7000 [7000] type=aor max_contacts=10 qualify_frequency=60 [7001] type=endpoint context=outgoing disallow=all allow=g722,alaw,ulaw transport=transport-udp auth=auth7001 aors=7001 direct_media = no disable_direct_media_on_nat = yes [auth7001] type=auth auth_type=userpass password=x username=7001 [7001] type=aor max_contacts=10 qualify_frequency=60 Am 07.05.2014 07:35, schrieb Rainer Piper: that's funny I recompiled asterisk without bridge_native_rtp.so to force asterisk to go to simple_bridge and not to native_bridge... !!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu Am 07.05.2014 07:11, schrieb Rainer Piper: PS. if I configure both extension 7000 and 7001 to, disallow=all allow=alaw or disallow=all allow=g722 everything is fine. as long as the allowed codec is equal in both extensions. Am 07.05.2014 07:00, schrieb Rainer Piper: Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de -- -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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
