Chris Hertz
2005-Jan-18 14:31 UTC
[Asterisk-Users] Asterisk and IAX softphone (firefly) problem/question
Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly immediately picks up the call without warning the receiver. Then oddly enough while the call is already connected, still gives the option of accepting or rejecting the call. Then after 20 seconds (as specified in extensions.conf), if the receiver hasn't accepted the call on Firefly Asterix pulls the call back and dumps it into voicemail. I have played with the settings, but can't seem to figure out why Firefly is automatically picking up the call. Below I have included relevant (or at least what I thought was relevant) code from extensions, iax and debug. Any help would be greatly appreciated. Thanks, Chris **************extensions.conf (snippet)********************* [default] exten => s,1,Wait,1 ; Wait a second, to create buffer for lag exten => s,1,DigitTimeout,5 ; Set Digit Timeout exten => s,2,ResponseTimeout,10 ; Set Response Timeout exten => s,3,BackGround(demo-congrats) ; Play a congratulatory message exten => s,4,BackGround(demo-instruct) ; Play some instructions ; extension 101 exten => 101,1,Playback(transfer,skip) exten => 101,2,Macro(stdexten,101,IAX2/test/101) [macro-stdexten]; ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device(s) to ring exten => s,1,Dial(${ARG2},20,tT) ; Dial(type/identifier,timeout,options,URL) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-NOANSWER,2,Goto(default,s,1) exten => s-BUSY,1,Voicemail(b${ARG1}) exten => s-BUSY,2,Goto(default,s,1) exten => _s-.,1,Goto(s-NOANSWER,1) exten => a,1,VoicemailMain(${ARG1}) **************iax.conf (snippet)*************************** [test] type=friend context=outgoing auth=md5 secret=xxxxx notransfer=1 host=dynamic allow=ilbc allow=gsm mailbox=101 qualify=no callerid="Test Person" <xxx xxx xxxx> ********************debug output from external call (snippet)************************** -- Executing Dial("IAX2/voicepulse-in-01@ipaddress/8", "IAX2/test|20") in new stack -- Called test -- Call accepted by <ipaddress> (format gsm) -- Format for call is gsm -- IAX2/test/5 is ringing -- Nobody picked up in 20000 ms -- Hungup 'IAX2/test/5' -- Executing Goto("IAX2/voicepulse-in-01@ipaddress/8", "s-NOANSWER|1") in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing VoiceMail("IAX2/voicepulse-in-01@ipaddress/8", "u101") in new stack
Mark
2005-Jan-19 00:38 UTC
[Asterisk-Users] Asterisk and IAX softphone (firefly)problem/question
I had exactly the same problem. I de-installed Firefly, pruned my WinXP registry of all of the settings, re-installed, and have not had the problem since. I can't explain why, but that was my experience... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Hertz Sent: Tuesday, January 18, 2005 2:32 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk and IAX softphone (firefly)problem/question Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly immediately picks up the call without warning the receiver. Then oddly enough while the call is already connected, still gives the option of accepting or rejecting the call. Then after 20 seconds (as specified in extensions.conf), if the receiver hasn't accepted the call on Firefly Asterix pulls the call back and dumps it into voicemail. I have played with the settings, but can't seem to figure out why Firefly is automatically picking up the call. Below I have included relevant (or at least what I thought was relevant) code from extensions, iax and debug. Any help would be greatly appreciated. Thanks, Chris **************extensions.conf (snippet)********************* [default] exten => s,1,Wait,1 ; Wait a second, to create buffer for lag exten => s,1,DigitTimeout,5 ; Set Digit Timeout exten => s,2,ResponseTimeout,10 ; Set Response Timeout exten => s,3,BackGround(demo-congrats) ; Play a congratulatory message exten => s,4,BackGround(demo-instruct) ; Play some instructions ; extension 101 exten => 101,1,Playback(transfer,skip) exten => 101,2,Macro(stdexten,101,IAX2/test/101) [macro-stdexten]; ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device(s) to ring exten => s,1,Dial(${ARG2},20,tT) ; Dial(type/identifier,timeout,options,URL) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-NOANSWER,2,Goto(default,s,1) exten => s-BUSY,1,Voicemail(b${ARG1}) exten => s-BUSY,2,Goto(default,s,1) exten => _s-.,1,Goto(s-NOANSWER,1) exten => a,1,VoicemailMain(${ARG1}) **************iax.conf (snippet)*************************** [test] type=friend context=outgoing auth=md5 secret=xxxxx notransfer=1 host=dynamic allow=ilbc allow=gsm mailbox=101 qualify=no callerid="Test Person" <xxx xxx xxxx> ********************debug output from external call (snippet)************************** -- Executing Dial("IAX2/voicepulse-in-01@ipaddress/8", "IAX2/test|20") in new stack -- Called test -- Call accepted by <ipaddress> (format gsm) -- Format for call is gsm -- IAX2/test/5 is ringing -- Nobody picked up in 20000 ms -- Hungup 'IAX2/test/5' -- Executing Goto("IAX2/voicepulse-in-01@ipaddress/8", "s-NOANSWER|1") in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing VoiceMail("IAX2/voicepulse-in-01@ipaddress/8", "u101") in new stack _______________________________________________ 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