[asterisk-users] Transfer call from analog telephone

[Daniel Bareiro]

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Hi all!

I'm trying to doing a transfer from an analog extension to a SIP
extension but until the moment I was not successful.

I was testing both the recall key and uncomment the following
lines in the features.conf file:

blindxfer => #1
atxfer => *2

verifying previously that the extension uses the arguments "tT" with the
Dial application and to include the context "featuremap" in the context
in which I have defined the extensions (internal).

The telephone of the end with which the conversation is staying listens
the tones to try doing the transfer, but Asterisk does not give the dial
tone after *2 / #1 or the recall key.

I copy my configuration files after to have reverted the changes. If some
other data is necessary, don't doubt in consulting to me. The lines that I
added to the configuration files created in the installation are those that
are underneath "DGB".

###################### /etc/asterisk/features.conf 

[general]
parkext => 700                  ; What extension to dial to park
parkpos => 701-720              ; What extensions to park calls on. These needs to be
                                ; numeric, as Asterisk starts from the start position
                                ; and increments with one for the next parked call.
context => parkedcalls          ; Which context parked calls are in
                                ; (default is 45 seconds)
                                ; when someone dials a parked call
                                ; or the Touch Monitor is activated/deactivated.
                                ; one of: parked, caller, both  (default is caller)
                                ; one of: callee, caller, both, no (default is both)
                                ; one of: callee, caller, both, no (default is no)
                                ; one of: callee, caller, both, no (default is no)
                                ; one of: callee, caller, both, no (default is no)
                                ; Defaults to 'first' available
                                ; as long as the class is not set on the channel directly
                                ; using Set(CHANNEL(musicclass)=whatever) in the dialplan

                                ; (default is 3 seconds)
                            ; feature activation  (default is 1000 ms)


[featuremap]

[applicationmap]

###################### /etc/asterisk/extensions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
IAXINFO=guest                                   ; IAXtel username/password
TRUNK=Zap/G2                                    ; Trunk interface
TRUNKMSD=1                                      ; MSD digits to strip (usually 1 or 0)

[default]

; DGB
[internal]
exten => _2xx,1,Dial(SIP/${EXTEN},15,tTm)
exten => _2xx,2,VoiceMail(${EXTEN}@voicemail)
exten => _2xx,3,Playback(vm-goodbye)
exten => _2xx,4,Hangup

exten => *98,1,Answer
exten => *98,2,Wait(1)
exten => *98,3,VoiceMailMain(${CALLERID}@voicemail)
exten => *98,4,Hangup

exten => *600,1,Answer
exten => *600,2,Playback(demo-echotest)
exten => *600,3,Echo
exten => *600,4,Playback(demo-echodone)
exten => *600,5,Hangup

exten => _9.,1,Dial(DAHDI/1/${EXTEN:1})
exten => _9.,2,Hangup

exten => 1010,1,Dial(DAHDI/2,15,tTm)
exten => 1010,2,Hangup

include => phones

[phones]
include => internal

[incoming]


exten => s,1,Dial(SIP/201,15,tTm)
exten => s,2,Hangup

###################### /etc/asterisk/chan_dahdi.conf

[trunkgroups]

[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300              ; Atlas seems to use long (250ms) winks

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1

immediate=no
busydetect=yes

; DGB
language=es
defaultzone=es
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
inmediate=no

context=phones
signalling=fxo_ks
channel => 2       ; Telephone attached to port 2
context=incoming
signalling=fxs_ks  ; Use FXS signalling for an FXS channel
channel => 1       ; PSTN attached to port 1

######################


Which can be the problem or what configuration can be lacking?

Thanks in avance.

Regards,
Daniel

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