Hi,
i’m new to asterisks…i have created a basic auto dail service which when moved to outgoing i don’t see any auto calls being generated.
I have created a directory in outgoing folder for logs.is tis the cause for auto dail failing?
.call file contains
Channel: PJSIP/9091@trunk2
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: from-XXXXXX
Extension: 10
Thanks in Advance…
I’m not sure if creating directory under outgoing (I presume /var/spool/asterisk/outgoing) would result in some issues, however there are many things that could go wrong here,
To start with, do you see anything in Asterisk CLI when you place call file?
–Satish Barot
there is nothing mentioned in asterisk CLI when i place the call file in outgoing also the call file remained in the same location and was not processed.
What’s your Asterisk version? Did you try the same using AMI instead of callfile to see if it makes any difference?
Collect the debug information as mentioned on this link https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information and check if you see anything unusual in your logs.
Thanks for your response.However after restarting asterisk it started picking .call files from outgoing.
As part of working on Asterisk, I have to run 2 .call files. One gets called immediately whereas for the other sip phone I see the below error.
– Attempting call on PJSIP/9001@asterisk2 for application Playback(/var/lib/asterisk/sounds/Digest) (Retry 3)
– Attempting call on PJSIP/7001@asterisk2 for application Playback(/var/lib/asterisk/sounds/Religious) (Retry 3)
[Mar 31 15:46:01] WARNING[13397]: res_pjsip_sdp_rtp.c:1165 create_outgoing_sdp_stream: Unable to get rtp codec payload code for testlaw
[Mar 31 15:46:01] WARNING[13397]: res_pjsip_sdp_rtp.c:1165 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[Mar 31 15:46:01] WARNING[13397]: res_pjsip_sdp_rtp.c:1165 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[Mar 31 15:46:01] WARNING[13397]: res_pjsip_sdp_rtp.c:1165 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[Mar 31 15:46:01] WARNING[13397]: res_pjsip_sdp_rtp.c:1165 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
– Called 9001@asterisk2
– Called 7001@asterisk2
[Mar 31 15:46:01] WARNING[13397]: res_pjsip_sdp_rtp.c:1165 create_outgoing_sdp_stream: Unable to get rtp codec payload code for testlaw
[Mar 31 15:46:01] WARNING[13397]: res_pjsip_sdp_rtp.c:1165 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[Mar 31 15:46:01] WARNING[13397]: res_pjsip_sdp_rtp.c:1165 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[Mar 31 15:46:01] WARNING[13397]: res_pjsip_sdp_rtp.c:1165 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[Mar 31 15:46:01] WARNING[13397]: res_pjsip_sdp_rtp.c:1165 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk
[Mar 31 15:46:01] NOTICE[13719]: pbx_spool.c:413 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)
[Mar 31 15:46:01] NOTICE[13719]: pbx_spool.c:416 attempt_thread: Queued call to PJSIP/7001@asterisk2 expired without completion after 2 attempts
– PJSIP/asterisk2-0000010b is ringing
– PJSIP/asterisk2-0000010b answered
> Launching Playback(/var/lib/asterisk/sounds/ Digest) on PJSIP/asterisk2-0000010b
– <PJSIP/asterisk2-0000010b> Playing ‘/var/lib/asterisk/sounds/ Digest.gsm’ (language ‘en’)
> 0x91498f0 – Probation passed - setting RTP source address to 192.168.1.239:18292
[Mar 31 15:46:13] NOTICE[13720]: pbx_spool.c:426 attempt_thread: Call completed to PJSIP/9001@asterisk2