[Asterisk-bsd] Zap on hold problem
Diego Valencia
dvalencia at powervt.com.ar
Tue Mar 7 12:00:38 MST 2006
Sorry, I forgot include the "internos" context of the extensions.conf:
[internos]
exten => 233,1,Macro(stdexten,233,SIP/233)
exten => 240,1,Macro(stdexten,240,SIP/240)
----- Original Message -----
From: "Diego Valencia" <dvalencia at powervt.com.ar>
To: "Asterisk on BSD discussion" <asterisk-bsd at lists.digium.com>
Sent: Tuesday, March 07, 2006 3:14 PM
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
> mailbox=233 at default,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
> mailbox=233 at default,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" <marios at comand.net>
> To: "'Asterisk on BSD discussion'" <asterisk-bsd at lists.digium.com>
> 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: asterisk-bsd-bounces at lists.digium.com
> [mailto:asterisk-bsd-bounces at lists.digium.com] 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" <oej at edvina.net>
> To: "Asterisk on BSD discussion" <asterisk-bsd at lists.digium.com>
> Cc: "Olle E Johansson" <oej at edvina.net>
> 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
>> _______________________________________________
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>> Asterisk-BSD at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-bsd
>
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