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.
Related
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
--------------------------------------------------------------------------
I am trying to do a stress test on the asterisk server.
I have created multiple accounts that call each other.
For example
1001 calls -----> 1002
1003 calls -----> 1004
1005 calls -----> 1006
Somehow when the server gets to 64 active calls it keeps saying that there is no active port.
And I can't make any more calls.
The error log;
Choices:
0 For current dialog.
-1 All 0 buddies in buddy list
[1 - 0] Select from buddy list
URL An URL
<Enter> Empty input (or 'q') to cancel
Make call: 07:40:28.584 pjsua_call.c !Making call with acc #1 to sip:1006#127.0.0.1:25060
07:40:28.584 pjsua_aud.c .Set sound device: capture=-99, playback=-99
07:40:28.584 pjsua_aud.c ..Setting null sound device..
07:40:28.584 pjsua_app.c ...Turning sound device ON
07:40:28.584 pjsua_aud.c ...Opening null sound device..
07:40:28.584 pjsua_media.c .Call 0: initializing media..
07:40:28.584 pjsua_media.c ..Unable to find appropriate RTP/RTCP ports combination
07:40:28.584 pjsua_media.c ..Unable to create RTP/RTCP socket: Address already in use [status=120098]
07:40:28.584 pjsua_media.c ..Error creating media transport: Address already in use
07:40:28.584 pjsua_call.c .Error initializing media channel: Address already in use [status=120098]
07:40:28.584 pjsua_media.c .Call 0: deinitializing media..
>>> >>>>
Account list:
[ 0] <sip:172.31.31.91:30404>: does not register
Online status: Online
*[ 1] sip:1005#172.31.31.91: 200/OK (expires=219)
Online status: Online
Buddy list:
-none-
Anybody a clue why this keeps happening.
I have also created a script that registers 5000 users, all with different ports and this works perfectly without any problems. Would it be possible that there is somewhere a limit function?
Likly you have limited number of ports allowed in rtp.conf file
Please stop spam same questions, asterisk for sure can handle thousands of calls, check your setup(embeded?)/config.
I am using php version 5.3.10 and asterisk version 1.8.22.0.
I am registering one customer of a2billing in softphone and dialing one number.
In asterisk i am getting below result:
<SIP/myip-0000004c>AGI Tx >> 200 result=1
<SIP/myip-0000004c>AGI Rx << Connection failed
<SIP/myip-0000004c>AGI Tx >> 510 Invalid or unknown command
[Dec 30 07:59:16] ERROR[28331]: utils.c:1343 ast_carefulwrite: write() returned error: Broken pipe
[Dec 30 07:59:16] ERROR[28331]: utils.c:1343 ast_carefulwrite: write() returned error: Broken pipe
-- <SIP/myip-0000004c>AGI Script a2billing.php completed, returning 0
Does anybody have any idea what is the issue?
I am getting correct credential in AGI when it is trying to connect and using those credential i can connect in mysql but from CLI> i am getting connection failed error.
Thanks in advance.
You need to install php5-adodb and libphp-adodb.
I am applying an auto dial in asterisk using .call file
My a.call
Channel: DAHDI/g0/09*********
MaxRetries: 1
RetryTime: 600
WaitTime: 30
Context: outgoing
Extension: 10
Priority: 1
My problem is that every time above number is called by same number means even if i change the dialled number(receiver number above) the caller number is same.How can i set the caller number in an outgoing call?
Thanks in advance.
You can use
Channel: DAHDI/g0/09*********
MaxRetries: 1
RetryTime: 600
WaitTime: 30
Context: outgoing
Extension: 10
Priority: 1
Callerid: 12345
Note, dahdi g0 have be digital trunk and ALLOW change of callerid.
You just add the parameter on your .call file
Callerid: <your_callerid>
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