I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank you, Alex Fedorov
albor@ipeya.com wrote:> > I am trying implement two-stage dialing. > Scenario is following: > 1. * Dials SIP agent > 2. SIP agent answer the phone and provide dial tone > 3. * Sends DTMF string > 4. "Bridge" channel with calling party > I thought that something like: > exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) > exten => _2XX,3,Wait,1 > exten => _2XX,4,SendDTMF($DTMF_DIGITS)You can do "2 stage dialing" with a call file. This may be a hack, but it currently works. I feel it is backwards from all the samples I saw since it rings the Agent's SIP phone first. From what I see, call files connect the Channel first, then run the Context/Exten/Priority after it's picked up. -------- #Two stage dial call file #Rings the Channel first the connects the "Extension". #agent's SIP phone, like SIP/agent007 Channel: SIP/xws108 #Use a context that can dialout. #Probably whatever the agent's phone is set as Context: internal #Phone number to ring....You may need a '9' prefix #To Asterisk, Extensions don't have to be internal only... Extension: 5551212 #use '1' unless you know what you're doing Priority: 1 #If your telephone interface sends CallerID, be sure to set it. Callerid: 888-555-1212 --------- Make this file and then copy it into /var/spool/asterisk/outgoing. Don't try to make the files in that directory, because Asterisk may read it before your script is done writing the file. This may make Asterisk mad. P.S. You can also do this with the Manager Interface, I lifted enough perl code to make this work for me using a "Click to Dial" script that sucks in the phone number from a web interface. http://www.azxws.com/asterisk Tony Wasson
Hi, This is not gonna work, is it? Is there such thing as Dial_but_not_connect_? I am trying to do the same thing but don't know how to accomplish this. If you've or anyone here figured out, please let me know. Thank you very much, Ron albor@ipeya.com wrote: I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank you, Alex Fedorov _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040403/5f6e5c88/attachment.htm
-----Original Message----- From: Ron McMillin [SMTP:sipnow@sbcglobal.net] Sent: Saturday, April 03, 2004 6:33 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] two-stage dialing Hi, This is not gonna work, is it? Is there such thing as Dial_but_not_connect_? I am trying to do the same thing but don't know how to accomplish this. If you've or anyone here figured out, please let me know. Thank you very much, Ron albor@ipeya.com wrote: I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank you, Alex Fedorov _______________________________________________ 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 << File: ATT00083.htm >>