FW: [Asterisk-Users] Retrieving dtmf, passing to shell and getting the result
Title: FW: [Asterisk-Users] Retrieving dtmf, passing to shell and getting the result John thanks for the help. When I change my plan to this and then dial 2 it gives me a busy signal. When troubleshooting I added an exten = 2,1 Ringing (just as a check) it rang and went straight to busy. On the console I got: --Executing Ringing(SIP/-00816800, ) in new stack ==Spawn extension(default, 2, 2) exited non-zero on SIP/-00816800 Any ideas? Jane Jane, try this exten = 2,2,read (firstnumber,enter-first,5) exten = 2,3,read (secondnumber,enter-second,2) exten = 2,4,system(/usr/local/Scripts/CheckNumbers.sh ${firstnumber} ${secondnumber}) I believe it is the syntax that is holding you back. John M Original Post I have my asterisk server up and running on OS X and now need to add the capability to play a sound file asking for a 5 digit number, play another message asking for a 2 digit number, pass these variables to a shell script, and get the result. I have tried a number of different scenarios but they are not working. I have read through the wiki, past posts, and numerous websites. The sound files are enter-first enter-second The shell script is CheckNumbers.sh exten = 2,2,get_data (enter-first,1,5) exten = 2,3,get_data (enter-second,1,2) exten = 2,4,system(/usr/local/Scripts/CheckNumbers.sh ${firstnumber, secondnumber) I really appreciate your help! Jane ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Retrieving dtmf, passing to shell, and getting the result
Title: Retrieving dtmf, passing to shell, and getting the result I have my asterisk server up and running on OS X and now need to add the capability to play a sound file asking for a 5 digit number, play another message asking for a 2 digit number, pass these variables to a shell script, and get the result. I have tried a number of different scenarios but they are not working. I have read through the wiki, past posts, and numerous websites. The sound files are enter-first enter-second The shell script is CheckNumbers.sh exten = 2,2,get_data (enter-first,1,5) exten = 2,3,get_data (enter-second,1,2) exten = 2,4,system(/usr/local/Scripts/CheckNumbers.sh ${firstnumber, secondnumber) I really appreciate your help! Jane ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3k answers then immediate busy signal
Title: Sipura 3k answers then immediate busy signal I have a sipura 3000 that I am using just to send calls to my mac asterisk server. When you call the phone it rings, answers, and then goes right to a busy signal. Any ideas? Thanks for your help! Jane At the console in verbose mode I get: *CLI DEBUG[8501248]: File chan_sip.c, Line 663 (create_addr): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found *CLI DEBUG[8501248]: File chan_sip.c, Line 3898 (check_user): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Found DEBUG[8501248]: File chan_sip.c, Line 3898 (check_user): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 4950 (handle_request): Check for res for 400 DEBUG[8501248]: File chan_sip.c, Line 980 (find_user): Call from user '400' is 1 out of 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 102: Not Found DEBUG[8501248]: File chan_sip.c, Line 663 (create_addr): Setting NAT on RTP to 0 DEBUG[8501248]: File chan_sip.c, Line 554 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users