how to transfer a outgoing zap call in ringing

abhasbajpai

Joined
Nov 12, 2008
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#1
I want to know how to transfer a outgoing zap call in ringing state to a sip extensions

I have a setup a elastix system with pri and its running fine except this operator problem (which incidentally is not there in plane asterisk installation)

Here operator calls a pstn no hears the ringing tone then hits the transfer button and transfer the call to sip ext.
This was happening in plain vanilla asterisk installation but not in asterisk. I tried with polycom and grandstream phone
I have enable the Tt and Tt in asterisk dial command option and in dial trunk in genral settings
Is there some thing I am lacking

I am giving the my sip.conf & extensions.conf file of working plane asterisk installation

Extensions.conf

[general]

static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp
TRUNK=Zap/g1
TRUNK=Zap/g2

[demo]
include => default


[local]
include => default

[default]
exten =>8600,1,Meetme,8600
exten => 8601,1,Meetme,8601

exten => _2.,1,Dial(SIP/${EXTEN})
exten => _4.,1,Dial(SIP/${EXTEN})
exten => _7.,1,Dial(SIP/${EXTEN})

; For dial pad
exten => _9.,1,Answer
exten => _9.,2,Dial(Zap/g1/${EXTEN:1})
exten => _9.,3,Hangup

exten => *97,1,Voicemailmain(${CALLERID(num)})
exten => *98,1,Voicemailmain()

exten => 1710,1,Ringing
exten => 1710,2,Dial(SIP/710${EXTEN:4}|20|t)
exten => 1710,3,Hangup


sip.conf


[general]
context=default
port=5060
bindport=5080
bindaddr=0.0.0.0
srvlookup=yes

rtptimeout=3600
rtpholdtimeout=3600

disallow=all
allow=ulaw

[elastix]
type=peer
host=192.168.2.44
;fromdomain=202.54.112.194
fromuser=ast
;disallow=all
allow=all
;allow=ulaw
username=ast
secret=ast123
context=from-elastix






[710]
type=friend
username=710
;secret=101
callerid= <1123001710>
nat=yes
host=dynamic
port=5060
dtmfmode=rfc2833
canreinvite=no
context=default
disallow=all
allow=ulaw
allow=alaw
 

rafael

Joined
May 14, 2007
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#2
Here operator calls a pstn no hears the ringing tone then hits the transfer button and transfer the call to sip ext.
This was happening in plain vanilla asterisk installation but not in asterisk. I tried with polycom and grandstream phone
I have enable the Tt and Tt in asterisk dial command option and in dial trunk in genral settings
Is there some thing I am lacking
I guess it does not work on Elastix. I just did some testing on an estandard Elastix 1.3 and it worked for us.
 

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