similar to: RTCP + Stasis causing high memory consumption

2023-11-08 18:14:13] ERROR[571246][C-000017e2] : Got 19 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() # 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref() # 2: [0x58e660] asterisk stasis_cache.c:824 update_create() # 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk

HI! I'm trying to compile asterisk 13.8.2+ on openSUSE Linux but it fails. It seems file ./main/libasteriskssl.so.1 is present when it fails. Building 13.7.2 works without any problem. It fails since 13.8.0. $ ./bootstrap.sh $ ./configure $ make menuselect.makeopts;menuselect/menuselect --enable chan_ooh323 $ make .. failure (see message below) Any hint is appreciated. Thanks in advance.

Hi, While compiling 1.8.8.1, I met the following error: [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o

Hello all, mmm a bit embarrassing about not having a clue as to why we're getting this error on make of 1.8.3 [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o

Hello, we just installed the latest 13 and 16 version of asterisk and face problem on incoming calls: they are ended like in Asterisk 16 [2020-02-04 19:19:48] ERROR[3768][C-00000001]: stasis_bridges.c:199 bridge_topics_init: Bridge id initialization required [2020-02-04 19:19:48] WARNING[3768][C-00000001]: bridge.c:809 bridge_base_init: Bridge da3bd3d1-cdea-4a05-8b3d-0ded8c561c5f: Could not

Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip

I have a plain text file, ASCII, unix line breaks. 1 single line, and all that is in it is the word "radio". Here's some test dialplan: exten => 5,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN}) same => n,Set(feature=${FILE(/home/test/feature-1.txt,0,1,l,u)}) same => n,Verbose(${feature}) same => n,Set(featurefile=/home/test/feature-1.txt) same =>

Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init

Hello. I tryto use Statis at my dialplan to run my app (a) When Statis is running from making call ( I dial from softphone some exten and run dialplan context where call Statis(MyApp)) Asterisk responsed: ERROR[61517][C-00000019]: res_stasis.c:852 stasis_app_exec: Stasis app 'MyApp' not registered How I must Register MyApp -------------- next part -------------- An HTML attachment was

Hi, I'm writing a node.js backend to pass events via a websocket to a CRM. Basically what I want to do is notice when things happen (i.e. new channel, new bridge etc) without sending the channels to the Stasis app. The channels I'm interested in are agents who are in a queue only because they are in a realtime MySQL database for the queue_member_table. There doesn't appear to be a

On Thu, Aug 6, 2020 at 1:52 PM Dan Cropp <dan at amtelco.com> wrote: > I understand how to control the first local channel, but an having trouble > getting the second local channel to enter stasis. > > > > I setup have the following extensions.conf to handle 1000 (basically had > it setup so if first stasis not there try second, but believe second > channel never

I'm attempting to find where my extra long DTMF Tones are coming from. I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway. I'm pressing 4 to select a menu and everything is fine. [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin passthrough

Hi All, I'm new to Asterisk and I'm trying to manage the calls using the rest API. I want to play a media file after 5 minutes the call started, how can I move the call to a Stasis application using the HTTP API in order to play the media using play API( https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Channels+REST+API#Asterisk16ChannelsRESTAPI-play )? Thanks, Tripon Alexandru

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

On Tue, 18 Oct 2016, Pete Mundy wrote: > If you want to know what is _really_ in that file (including all > invisible characters and anything else that wc etc might not count), > pipe it through 'hexdump'. > > cat?/home/test/feature-1.txt | hexdump Or just: hexdump /home/test/feature-1.txt -- Thanks in advance,

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100

Hello All, I'm using Asterisk 16.8.0 on a Centos 7 box. Previously 16.5.0, But recently upgraded to attempt to resolve this issue. Using bundled PJSIP. The PBX is using mysql realtime for most functions. The Mysql server is on the same lan as the asterisk box. As more users have been moved to this box. It's become unstable. Randomly, I'll start seeing "WARNING[12667]

Hello, I'm load testing a new Asterisk 13 system (Debian Stretch, packaged asterisk). One system writes CDR though an ODBC connection to a local Postgres database over the LAN. When sending 50 new calls per second with SIPp, I'm seeing one system outputs : taskprocessor.c: The 'subm:cdr_engine-00000003' task processor queue reached 5000 scheduled tasks again. This [1] thread

examples of "interesting" information like ICE result and howto make "minimal" configuration of pjproject.conf i.e. for  debugging app_queue.so core set debug 5 app_queue.so for debugging RTP core set debug 10 rtp_engine core set debug 10 res_rtp_asterisk rtp set debug on logger.conf rtp => debug,verbose(5) so i mean in

Good afternoon list. I am experiencing a problem with the CDR and callfiles. What is happening is this: When generating a call with a callfile, everything works perfectly, but the CDR is recorded in the table when they answer the call destination. The field disposition is being recorded correctly, but the duration field is marked with the ring time and billsec is marked with 0. This just happens