Call is ringing only please help me ring actual call

[pmgawande]

when i do the web call is reached to the asterisk and it ringing only but not actual ring the sales person’s mobile phone. when i debug the asterisk i find the following error log

– Executing [456456@vtiger_inbound:1] Set(“SIP/gsm123123-00000000”, “CALLERID(num)=+91XXXXXXXXXX”) in new stack
– Executing [456456@vtiger_inbound:2] AGI(“SIP/gsm123123-00000000”, “agi://0.0.0.0/incoming.agi”) in new stack
– AGI Script Executing Application: (Monitor) Options: (wav,/usr/local/VtigerConnector/storage/b09ee50dff14e31262e530669,m)
[2019-12-20 05:11:29] WARNING[14459][C-0000]: file.c:1230 ast_writefile: Unable to open file /usr/local/VtigerConnector/storage/b09ee50dff14e31262e530669-in.wav: Permission denied
[2019-12-20 05:11:29] WARNING[14459][C-00000000]: res_monitor.c:370 __ast_monitor_start: Could not create file /usr/local/VtigerConnector/storage/b09ee50dff14e31262e530669-in
[2019-12-20 05:11:29] WARNING[14459][C-00000000]: res_monitor.c:642 __ast_monitor_change_fname: Cannot change monitor filename of channel SIP/gsm123123-00000000 to /usr/local/VtigerConnector/storage/b09ee50dff14e31262e530669, monitoring not started
– AGI Script Executing Application: (Dial) Options: (SIP/gsm123123/88XXXXXXXX, 60)
== Using SIP RTP CoS mark 5
– Called SIP/gsm123123/88XXXXXXXX
– SIP/gsm123123-00000001 is making progress passing it to SIP/gsm123123-00000000
[2019-12-20 05:11:48] WARNING[14434]: db.c:332 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[2019-12-20 05:11:55] WARNING[14434]: chan_sip.c:4100 retrans_pkt: Timeout on 1690550185-318110118-1416081917 on non-critical invite transaction.
– Got SIP response 486 “Busy Here” back from 103.44.110.172:15728
– SIP/gsm123123-00000001 is busy
== Everyone is busy/congested at this time (1:1/0/0)
– <SIP/gsm123123-00000000>AGI Script agi://0.0.0.0/incoming.agi completed, returning 0
– Executing [456456@vtiger_inbound:3] Hangup(“SIP/gsm123123-00000000”, “”) in new stack
== Spawn extension (vtiger_inbound, 456456, 3) exited non-zero on ‘SIP/gsm123123-00000000’
[2019-12-20 05:11:57] ERROR[14459][C-00000000]: cdr_csv.c:314 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
== Using SIP RTP CoS mark 5
[2019-12-20 05:12:00] NOTICE[14434][C-00000003]: chan_sip.c:25902 handle_request_invite: Call from ‘1010’ (45.143.220.92:5078) to extension ‘01146340683420’ rejected because extension not found in context ‘testing’.
[2019-12-20 05:12:10] WARNING[14434]: chan_sip.c:4100 retrans_pkt: Timeout on 52950715-1677994787-81XXXXXXXX on non-critical invite transaction.
[2019-12-20 05:12:20] WARNING[14434]: db.c:332 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[2019-12-20 05:12:32] WARNING[14434]: chan_sip.c:4038 retrans_pkt: Retransmission timeout reached on transmission a6b4c55bc266a891ef2fab4fbdbd1528 for seqno 1 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[2019-12-20 05:12:41] WARNING[14434]: chan_sip.c:4100 retrans_pkt: Timeout on 421440799-1109204043-286141043 on non-critical invite transaction.
[2019-12-20 05:12:52] WARNING[14434]: db.c:332 ast_db_put: Couldn’t execute statment: SQL logic error or missing database

[david551]

You have multiple problems in that log, and you haven’t provided the actual dialplan and the contents of the AGI script.

However, I think your issue is that ringing is being sent as early media. If the caller is local, you may be able to deal with this by calling the Progress() application before the Dial.

You should really fix the the explicitly reported errors before seeking support on one that isn’t reported in the logs.