![]() Under routing added dr_gateway with type 1 and added ip address of DID provider trunk. VERBOSE app_stack.With minor tweaks for "Debian Buster" I have installed all components, services starts with no error, I created few subscribers from opensipscp, under alias I added DID number pointing username of sip device, ![]() VERBOSE app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/201-0000018d’ VERBOSE res_agi.c: AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0 VERBOSE pbx.c: Executing :2] NoOp(“SIP/201-0000018d”, “HANGUP CAUSE: 34”) in new stack Tail -f /var/log/asterisk/full | grep 0000018d Im thinking its because i need to set a CID VERBOSE pbx.c: Spawn extension (from-internal, 155, 12) exited non-zero on ‘SIP/201-0000018d’ VERBOSE app_dial.c: Called SIP/voipfone-sip/ VERBOSE app_stack.c: Spawn extension (from-pstn, 155, 1) exited non-zero on ‘SIP/voipfone-sip-0000018e’ VERBOSE pbx.c: Executing :9] Set(“SIP/201-0000018d”, “_EMAILNOTIFICATION=FALSE”) in new stack VERBOSE pbx.c: Executing :8] Set(“SIP/201-0000018d”, “_CALLERIDNUMINTERNAL=201”) in new stack VERBOSE pbx.c: Executing :7] Set(“SIP/201-0000018d”, “_CALLERIDNAMEINTERNAL=robert wild”) in new stack VERBOSE pbx.c: Executing :6] Set(“SIP/201-0000018d”, “MOHCLASS=default”) in new stack VERBOSE pbx.c: Executing :5] Set(“SIP/201-0000018d”, “_ROUTENAME=out-by-voipfone”) in new stack ![]() VERBOSE pbx.c: Executing :1] NoOp(“SIP/201-0000018d”, “Outbound Recording Check from 201 to 155”) in new stack VERBOSE pbx.c: Executing :1] Macro(“SIP/201-0000018d”, “user-callerid,LIMIT,EXTERNAL,”) in new stack Tail -f /var/log/asterisk/full | grep 155 ::selection įont: 13px/20px normal Helvetica, Arial, sans-serif įont-family: Consolas, Monaco, Courier New, Courier, monospace Message: Cannot modify header information - headers already sent by (output started at /usr/local/stikked/Stikked-0.12.0/htdocs/system/core/Output.php:538) ![]()
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