This is strange. So you press line 1 again on eyeBeam and it doesn't get you back to the first call? Hmm. Let's try then the Asterisk transfer instead or the eybeam. In features.conf change ;blindxfer => #1 To blindxfer => #
In *CLI> reload res_features.so Make a call to Zap->eyeBeam Answer eyeBeam and press # You should hear "Transferring" Enter another extension once successful transfer eyeBeam will hangup If this works then there is no problem with asterisk and Zap. Usually on my eyeBeam I press line 2 enter a number (extension or a PSTN number) once the other extension answers then I press xfer and the two are connected. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Valencia Sent: Tuesday, March 07, 2006 1:15 PM To: Asterisk on BSD discussion Subject: Re: [Asterisk-bsd] Zap on hold problem Hi Marios, I don't have problem transfering sips, I only have problem when the call is coming form zap channel. There is a setting for zapata transfers? Theses are my conf: features.conf ; ; Sample Parking configuration ; [general] parkext => 700 ; What ext. to dial to park parkpos => 701-720 ; What extensions to park calls on context => parkedcalls ; Which context parked calls are in ;parkingtime => 45 ; Number of seconds a call can be parked for ; (default is 45 seconds) ;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call ;xfersound = beep ; to indicate an attended transfer is complete ;xferfailsound = beeperr ; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;findslot => next ; Continue to the 'next' parking space. Defaults to 'first' available pickupexten = 8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] ;blindxfer => #1 ; Blind transfer ;disconnect => *0 ; Disconnect ;automon => *1 ; One Touch Record ;atxfer => *2 ; Attended transfer [applicationmap] ;testfeature => #9,callee,Playback,tt-monkeys ;Play tt-monkeys to sip.conf [233] canreinvite=no username=233 type=friend context=nacionales secret=secret233 ;subscribecontext=trunklocal language=es host=dynamic [EMAIL PROTECTED],233 disallow=all allow=g729 allow=ulaw allow=alaw [240] canreinvite=no username=240 type=friend context=nacionales secret=secret240 ;subscribecontext=trunklocal language=es host=dynamic [EMAIL PROTECTED],233 disallow=all allow=g729 allow=ulaw allow=alaw extensions.conf: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device(s) to ring ; exten => s,1,Dial(${ARG2},30,t) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Dial(SIP/1222,30,) ; retorana a la consola exten => s-NOANSWER,2,Hangup ;exten => s-BUSY,1,MusicOnHold(ringbusy) ; If busy, send to voicemail w/ busy announce exten => s-BUSY,1,Hangup exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer [incomingzap] include => internos exten => s,1,Wait,1 ; Wait a second, just for fun ;exten => s,n,Set(SIP_CODEC=ulaw) exten => s,2,Answer ; Answer the line exten => s,3,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds exten => s,4,Set(TIMEOUT(response)=3) ; Set Response Timeout to 10 seconds exten => s,5,Set(LANGUAGE()=es) ; Set language to french exten => s,6(restart),BackGround(welcome) ; Play a congratulatory message exten => s,7,WaitExten ; Wait for an extension to be dialed. exten => s,8,Dial(SIP/232,30 zapata.conf [channels] faxdetect=incoming hanguponpolarityswitch=yes busydetect=yes busycount=4 immediate => no transfer => yes cancallforward => yes threewaycalling => yes callreturn => yes usecallerid=yes hidecallerid=no group => 1 context => incomingzap signalling => fxs_ks amaflags => documentation echocancel=yes ;Cancela el echo producido por las lineas análogas echocancelwhenbridged=yes echotraining=yes channel => 1-2 ------------------------------ Call flow: -- Starting simple switch on 'Zap/2-1' Mar 5 13:46:41 NOTICE[3477]: chan_zap.c:6063 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait("Zap/2-1", "1") in new stack -- Executing Answer("Zap/2-1", "") in new stack -- Executing Set("Zap/2-1", "TIMEOUT(digit)=5") in new stack -- Digit timeout set to 5 -- Executing Set("Zap/2-1", "TIMEOUT(response)=3") in new stack -- Response timeout set to 3 -- Executing Set("Zap/2-1", "LANGUAGE()=es") in new stack -- Executing BackGround("Zap/2-1", "welcome") in new stack -- Playing 'welcome' (language 'es') == CDR updated on Zap/2-1 -- Executing Macro("Zap/2-1", "stdexten|233|SIP/233") in new stack (233 is eyebeam) -- Executing Dial("Zap/2-1", "SIP/233|30|t") in new stack -- Called 233 -- SIP/233-7aa8 is ringing -- SIP/233-7aa8 answered Zap/2-1 -------------> I press "line 2" button on eyebeam to call to other extension -- Started music on hold, class 'default', on Zap/2-1 ---------> MOH on ZAP At this point the caller (PSTN) is on MOH, but I can't return to call 1 to transfer it. After a few minutes the eyebeam says "Failed to place call on hold" Thanks Diego ----- Original Message ----- From: "Marios Andreou" <[EMAIL PROTECTED]> To: "'Asterisk on BSD discussion'" <[email protected]> Sent: Tuesday, March 07, 2006 12:54 PM Subject: RE: [Asterisk-bsd] Zap on hold problem I'm using eyeBeam and I never had a problem with HOLD and Transfer with asterisk. It might be something with your extensions.conf setup. Do you have the 't' or 'T' option in the Dial from the ZAP to the SIP ? Do you have enabled transfers in the features ? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Valencia Sent: Tuesday, March 07, 2006 9:56 AM To: Asterisk on BSD discussion Cc: Olle E Johansson Subject: Re: [Asterisk-bsd] Zap on hold problem Hi Olle, thanks for you reply. Can you help me about my problem? I can't transfer the call when it is coming from zap channel. I want to do this: PSTN ---> ZAP ----> SIP ----transfer---> SIP Is it posible? When I press hold button, on the pstn side, starts MOH, but I can't return to the previous call any more. The eyebeam says "Failed to place call on hold". I see that the UA recieves "not found" from asterisk when it sends the "on hold" INVITE. I was searching on the net and I can't find a user with the same problem. :o( I guess that I'm doing something wrong. Thanks for any help. BR Diego ----- Original Message ----- From: "Olle E Johansson" <[EMAIL PROTECTED]> To: "Asterisk on BSD discussion" <[email protected]> Cc: "Olle E Johansson" <[EMAIL PROTECTED]> Sent: Monday, March 06, 2006 5:23 PM Subject: Re: [Asterisk-bsd] Zap on hold problem > > 6 mar 2006 kl. 20.47 skrev Diego Valencia: > >> Hi, anybody knows if is normal the "Ignoring this INVITE request"?: >> The call is incoming from zap channel, this invite is when I put the >> call on hold, and the UA does not get a response. > This means that we are getting a repeated transmission of an INVITE that > we already have and are processing. The second one will be ignored. > > /O > _______________________________________________ > Asterisk-BSD mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-bsd _______________________________________________ Asterisk-BSD mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-bsd _______________________________________________ Asterisk-BSD mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-bsd _______________________________________________ Asterisk-BSD mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-bsd _______________________________________________ Asterisk-BSD mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-bsd
