I submitted a job via slurm. The job ran for 12 hours and was working as expected. Then I got Data unpack would read past end of buffer in file util/show_help.c at line 501. It is usual for me to get errors like ORTE has lost communication with a remote daemon but I usually get this in the beginning of the job. It is annoying but still does not cause as much time loss as getting error after 12 hours. Is there a quick fix for this? Open MPI version is 4.0.1.
--------------------------------------------------------------------------
By default, for Open MPI 4.0 and later, infiniband ports on a device
are not used by default. The intent is to use UCX for these devices.
You can override this policy by setting the btl_openib_allow_ib MCA parameter
to true.
Local host: barbun40
Local adapter: mlx5_0
Local port: 1
--------------------------------------------------------------------------
--------------------------------------------------------------------------
WARNING: There was an error initializing an OpenFabrics device.
Local host: barbun40
Local device: mlx5_0
--------------------------------------------------------------------------
[barbun21.yonetim:48390] [[15284,0],0] ORTE_ERROR_LOG: Data unpack would read past end of buffer in
file util/show_help.c at line 501
[barbun21.yonetim:48390] 127 more processes have sent help message help-mpi-btl-openib.txt / ib port
not selected
[barbun21.yonetim:48390] Set MCA parameter "orte_base_help_aggregate" to 0 to see all help / error
messages
[barbun21.yonetim:48390] 126 more processes have sent help message help-mpi-btl-openib.txt / error in
device init
--------------------------------------------------------------------------
Primary job terminated normally, but 1 process returned
a non-zero exit code. Per user-direction, the job has been aborted.
--------------------------------------------------------------------------
--------------------------------------------------------------------------
An MPI communication peer process has unexpectedly disconnected. This
usually indicates a failure in the peer process (e.g., a crash or
otherwise exiting without calling MPI_FINALIZE first).
Although this local MPI process will likely now behave unpredictably
(it may even hang or crash), the root cause of this problem is the
failure of the peer -- that is what you need to investigate. For
example, there may be a core file that you can examine. More
generally: such peer hangups are frequently caused by application bugs
or other external events.
Local host: barbun64
Local PID: 252415
Peer host: barbun39
--------------------------------------------------------------------------
--------------------------------------------------------------------------
mpirun detected that one or more processes exited with non-zero status, thus causing
the job to be terminated. The first process to do so was:
Process name: [[15284,1],35]
Exit code: 9
--------------------------------------------------------------------------
Consider the following asterisk dialplan. After the last sound file is played it hangs up immediately and the waitexten timeout parameter seems to be ignored. The behaviour I am trying to achieve is to wait a given number of seconds after the last sound file is completed to get a response and then hang up. The last sound file says "Dial * to hear these options again". If I set the waitexten timeout to say 60, it hangs up before the sound files are played. A shorter time allows them all to be played but then it hangs up immediately. Any suggestions for handling this a better way are welcome
[mainmenu]
exten => s,1,Wait(0.25)
same => 2,Answer()
same => 3,Background(/srv/asterisk/sounds/optionslist)
same => n,Background(/srv/asterisk/sounds/dial2cs)
same => n,Background(/srv/asterisk/sounds/dial3ma)
same => n,Background(/srv/asterisk/sounds/dial4ac)
same => n,Background(/srv/asterisk/sounds/dial0)
same => n,Background(/srv/asterisk/sounds/dialstar)
same => n,WaitExten(20)
exten => 2,1,Goto(cs,2,1)
exten => *,1,Goto(s,3)
console output
== Using SIP RTP CoS mark 5
-- Executing [+12345#public:1] Goto("SIP/xxx.pstn.twilio.com-00000044", "mainmenu,s,1") in new stack
-- Goto (mainmenu,s,1)
-- Executing [s#mainmenu:1] Wait("SIP/xxx.pstn.twilio.com-00000044", "0.25") in new stack
-- Executing [s#mainmenu:4] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial2cs") in new stack
-- Executing [s#mainmenu:5] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial3ma") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dial3ma.slin' (language 'en')
-- Executing [s#mainmenu:6] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial4ac") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dial4ac.slin' (language 'en')
-- Executing [s#mainmenu:7] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial0") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dial0.slin' (language 'en')
-- Executing [s#mainmenu:8] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dialstar") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dialstar.slin' (language 'en')
[
[Aug 4 05:03:28] WARNING[2225]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission xxx#0.0.0.0 for seqno 5305 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[Aug 4 05:03:28] WARNING[2225]: chan_sip.c:4204 retrans_pkt: Hanging up call xxx#0.0.0.0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (mainmenu, s, 8) exited non-zero on 'SIP/xxx.pstn.twilio.com-00000044'
Asterisk 11.7.0~dfsg-1ubuntu1
I assume that you have a problem like in this topic Asterisk,SIP Retransmission timeout. Try to solve the problem with the NAT settings or firewall.
In this answer propose to set canreinvite=no in sip.conf.
I am currently working with freepbx version 2.11.038 using asterisk version 11.6. I am trying to set up an English Extension and a Spanish Extension ie when you ring one number you hit the ext 100049 which is for English and when you ring the other you hit ext 100050 which of for Spanish.
I need the voicemails of these extensions to be in the required language ie 100049 has English voice prompts and 100050 has Spanish voice prompts.
Currently when ringing both I get an English voice.
In the Settings->Advanced Sip Settings, I can set the language to say es. But then all the voice prompts are changed to Spanish on a global scale. I am looking to only do a single extension.
In Settings->Voicemail admin, I can set the various extensions voicemail language using a language code. The issue is that when a user calls both extensions they get the English voice prompts still. However if 100049 dial "*97" for voicemail they hear a password request in English, if 100050 dial "*97" for voicemail they are requested the password in Spanish. So this voicemail settings only seems to affect when a user dials into their own voicemail and not when an external calls hits their voicemail.
Here is my structure of the sound files
Here are my Extensions
Today I have installed the FreePBX language module, which seems to allow me to enter the extension language on the extension page rather than via the voicemail admin, but the result is the same, ext to "*97" language correct, external calls still gets default language.
Here are the log files for the calls made to ext 100049, the English Extension
-- Executing [vmx#macro-vm:10] NoOp("SIP/NodoProvicnial-000001cd", "Checking if ext 100049 is enabled: ") in new stack
-- Executing [vmx#macro-vm:11] GotoIf("SIP/NodoProvicnial-000001cd", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL#macro-vm:1] Macro("SIP/NodoProvicnial-000001cd","get-vmcontext,100049") in new stack
-- Executing [s#macro-get-vmcontext:1] Set("SIP/NodoProvicnial-000001cd", "VMCONTEXT=default") in new stack
-- Executing [s#macro-get-vmcontext:2] GotoIf("SIP/NodoProvicnial-000001cd","0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s#macro-get-vmcontext:300] NoOp("SIP/NodoProvicnial-000001cd","") in new stack
-- Executing [s-CHANUNAVAIL#macro-vm:2] VoiceMail("SIP/NodoProvicnial-000001cd", "100049#default,u") in new stack
-- <SIP/NodoProvicnial-000001cd> Playing 'vm-theperson.gsm' (language 'en')[2014-08-07 11:10:52] NOTICE[4184][C-000000d1]: channel.c:4259 __ast_read: Dropping incompatible voice frame on SIP/NodoProvicnial-000001cd of format g729 since our native format has changed to (alaw)
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/1.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/4.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'digits/9.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'vm-isunavail.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001cd> Playing 'vm-intro.gsm' (language 'en')
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'exten-vm'
== Spawn extension (from-did-direct, 100049, 2) exited non-zero on 'SIP/NodoProvicnial-000001cd'
-- Executing [h#from-did-direct:1] Macro("SIP/NodoProvicnial-000001cd", "hangupcall,") in new stack
-- Executing [s#macro-hangupcall:1] GotoIf("SIP/NodoProvicnial-000001cd", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s#macro-hangupcall:3] ExecIf("SIP/NodoProvicnial-000001cd", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s#macro-hangupcall:4] Hangup("SIP/NodoProvicnial-000001cd", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'hangupcall'
== Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/NodoProvicnial-000001cd'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/NodoProvicnial-000001cd
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'exten-vm'
== Spawn extension (from-did-direct, 100049, 2) exited non-zero on 'SIP/NodoProvicnial-000001cd'
-- Executing [h#from-did-direct:1] Macro("SIP/NodoProvicnial-000001cd", "hangupcall,") in new stack
-- Executing [s#macro-hangupcall:1] GotoIf("SIP/NodoProvicnial-000001cd", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s#macro-hangupcall:3] ExecIf("SIP/NodoProvicnial-000001cd", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s#macro-hangupcall:4] Hangup("SIP/NodoProvicnial-000001cd", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/NodoProvicnial-000001cd' in macro 'hangupcall'
== Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/NodoProvicnial-000001cd'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/NodoProvicnial-000001cd
Here are the logs for ext 100050 ie Spanish Extension:
-- Executing [vmx#macro-vm:10] NoOp("SIP/NodoProvicnial-000001d0", "Checking if ext 100050is enabled: ") in new stack
-- Executing [vmx#macro-vm:11] GotoIf("SIP/NodoProvicnial-000001d0", "1?s-NOANSWER,1") in new stack
-- Goto (macro-vm,s-NOANSWER,1)
-- Executing [s-NOANSWER#macro-vm:1] Macro("SIP/NodoProvicnial-000001d0", "get-vmcontext,100050") in new stack
-- Executing [s#macro-get-vmcontext:1] Set("SIP/NodoProvicnial-000001d0","VMCONTEXT=default") in new stack
-- Executing [s#macro-get-vmcontext:2] GotoIf("SIP/NodoProvicnial-000001d0", "0?00:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s#macro-get-vmcontext:300] NoOp("SIP/NodoProvicnial-000001d0", "") in new stack
-- Executing [s-NOANSWER#macro-vm:2] VoiceMail("SIP/NodoProvicnial-000001d0","100050#default,u") in new stack
-- <SIP/NodoProvicnial-000001d0> Playing 'vm-theperson.gsm' (language 'en')[2014-08-07 11:14:50] NOTICE[4191][C-000000d3]: channel.c:4259 __ast_read: Dropping incompatible voice frame on SIP/NodoProvicnial-000001d0 of format g729 since our native format has changed to (alaw)
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/1.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/5.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'vm-isunavail.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'vm-intro.gsm' (language 'en')
-- <SIP/NodoProvicnial-000001d0> Playing 'beep.gsm' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/100050/tmp/JKE7il format: wav49, 0x7fccb8064b98
-- x=1, open writing: /var/spool/asterisk/voicemail/default/100050/tmp/JKE7il format: gsm, 0x7fccb8060a78
-- x=2, open writing: /var/spool/asterisk/voicemail/default/100050/tmp/JKE7il format: wav, 0x7fccb807eac8
-- User hung up
I have asked this on the FreePBX forum also please see the following link.
FreePbx Forum Post
Any advice would be great as I am at a bit of a loss. Any other information I can provide please let me know.
You have download module called "Languages" via module admin and setup other language.
After that you can change language in dialplan, for example select in ivr on press 1 language es and go next destination as you set it.
Also you can change language in extensions tab, that will work for outbound calls from thoose extension.
I am trying to make a outgoing from an asterisk pbx using .call file but every time .call file is moved in outgoing folder my cli shows
[Jun 16 15:38:12] NOTICE[30435]: pbx_spool.c:372 attempt_thread: Call failed to go through, reason (1) Hangup
[Jun 16 15:38:12] NOTICE[30435]: pbx_spool.c:375 attempt_thread: Queued call to DAHDI/g0/09716927126 expired without completion after 0 attempts
-- Span 1: Channel 0/1 got hangup request, cause 16
-- Hungup 'DAHDI/i1/09711590094-103a'
[Jun 16 15:38:17] NOTICE[30434]: pbx_spool.c:372 attempt_thread: Call failed to go through, reason (1) Hangup
[Jun 16 15:38:17] NOTICE[30434]: pbx_spool.c:375 attempt_thread: Queued call to DAHDI/g0/09711590094 expired without completion after 0 attempts
-- Attempting call on DAHDI/g0/09711590094 for 4759509#outgoing1:1 (Retry 1)
-- Attempting call on DAHDI/g0/09716927126 for 4759509#outgoing1:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Requested transfer capability: 0x00 - SPEECH
-- Span 1: Channel 0/2 got hangup request, cause 31
-- Hungup 'DAHDI/i1/09716927126-103d'
my .call file
Channel: DAHDI/g0/09711590094
MaxRetries: 1
RetryTime: 600
WaitTime: 30
Context: outgoing1
Extension: 10
Priority: 1
The call could not be connected.Anybody knows what would be the possible reason for that?
Thanks in advance
This error mean you can't call as requested via dahdi/g0
Very likly you have configure correctly your dahdi card.
A user recently notified me that whenever they attempt to dial into a conference call at another company, the phone call would get dropped after 5 seconds or so. They also indicated that when the same number is called using a cell phone, there were no issues. I found the following entries in log file.
[May 4 11:58:20] VERBOSE[24063] app_dial.c: -- DAHDI/1-1 is ringing
[May 4 11:58:20] VERBOSE[24063] app_dial.c: -- DAHDI/1-1 answered SIP/145-00000005
[May 4 11:58:24] WARNING[24063] rtp.c: Don't know how to represent 'f'
[May 4 11:58:24] VERBOSE[24063] chan_dahdi.c: -- Redirecting DAHDI/1-1 to fax extension
[May 4 11:58:24] VERBOSE[24063] pbx.c: -- Executing [h#macro-dialout-trunk:1] Macro("SIP/145-00000005", "hangupcall,") in new stack
[May 4 11:58:24] VERBOSE[24063] pbx.c: -- Executing [s#macro-hangupcall:1] GotoIf("SIP/145-00000005", "1?theend") in new stack
I have not been able to determine a solution. Any insight or suggestions on solving this problem are appreciated.
(Using FreePBX v2.9; Asterisk v1.6.2.15.1; CentOS 5.5 (Final); Sangoma A102)
Try add into file
/etc/asterisk/sip_general_custom.conf
faxdetect=no
Also tried modiying chan_dahdi.conf, but that did not work.
Final solution was to modify these settings (changing from YES to NO) in /etc/wanrouter/wanpipe1.conf
TDMV_HW_DTMF = NO # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events from hardware