Hi Gireesh,

My apologies for the previous post. I didn't look into your mail so
clearly. Are you using transfer feature using pound key (#)?

Regards,
Kurian Thayil.

On Friday 25 March 2011 05:18 PM, Kurian Thayil wrote:
> Hi Gireesh,
>
> Post your features.conf file.
>
> Regards,
> Kurian Thayil.
>
> On Thursday 24 March 2011 08:46 AM, gireesh thenayil rajappan nair wrote:
>> Dear Joseph,
>>
>> pls tell me what can i edit features.conf sip.conf and other .conf
>> files for configuration ....what are the context's need to enter ..
>>
>> I got some configuration settings from
>>
>> http://forums.digium.com/viewtopic.php?f=1&t=4947&p=15317
>> <http://forums.digium.com/viewtopic.php?f=1&t=4947&p=15317>
>>
>> Im having problems getting the features.conf atxfer attended transfer
>> working.
>>
>> When caller1 calls caller 2 and caller 2 try to transfer the call it
>> is not happening..
>>
>>  asterisk -vvvvvvr   after that reload the system it is not working...
>>
>> extensions.conf
>>
>> exten => 601,1,Dial(SIP/601,30,rt)
>> exten => 601,2,VoiceMail,u1600
>> exten => 601,102,VoiceMail,b1600
>> exten => 601,103,HangUp()
>>
>> ;Nick
>> exten => 602,1,Dial(SIP/602,30,rt)
>> exten => 602,2,VoiceMail,u1600
>> exten => 602,102,VoiceMail,b1600
>> exten => 602,103,HangUp()
>>
>> features.conf
>> [featuremap]
>> blindxfer => #1 ; Blind transfer
>> disconnect => *0 ; Disconnect
>> ;automon => *1 ; One Touch Record
>> atxfer => *2 ; Attended transfer
>>
>>
>>
>> sip.conf
>>
>> allow=alaw
>> allow=h263
>> allow=h263p
>> dtfmode=rfc2833
>> canreinvite=no
>>
>>
>> [601]
>> type=friend
>> username=601
>> secret=12345
>> host=dynamic
>> context=office
>> callerid=user1<601>
>>
>> ;call-limit=1
>>
>> [602]
>> type=friend
>> username=602
>> secret=12345
>> nat=yes
>> host=dynamic
>> context=office
>> callerid=user2 <602>
>> call-limit=1
>>
>>
>> Pls communicate little deeper .....in to the configuration system...
>>
>>
>> Regards,
>>
>> Gireesh
>>
>> On Wed, Mar 23, 2011 at 6:19 PM, John Joseph <[email protected]
>> <mailto:[email protected]>> wrote:
>>
>>     Hi Gireesh
>>     Thanks for reply
>>     For trouble shooting you can give
>>     "asterisk -vvvvvvvvvvvvvvvvvvvvvvvr"
>>      at the command mode, this will give you all happening in verbose
>>     mode
>>     Now do call transfer and see the message coming on the terminal
>>     which you had given "asterisk -vvvvvvvvvvvvvvvvvvvvvvvr"
>>     Thanks
>>      Joseph John
>>
>>
>>     --- On Wed, 23/3/11, gireesh thenayil rajappan nair
>>     <[email protected] <mailto:[email protected]>> wrote:
>>
>>     > From: gireesh thenayil rajappan nair <[email protected]
>>     <mailto:[email protected]>>
>>     > Subject: Re: [ILUG-Cochin.org] Asterisk features.conf pls help
>>     > To: "This List discusses GNU/Linux & GNU, GPL Software"
>>     <[email protected] <mailto:[email protected]>>
>>     > Date: Wednesday, 23 March, 2011, 4:42 PM
>>     > Dear All,
>>     > vi /proc/version show's  Linux version 2.6.18-8.el5
>>     > ([email protected]
>>     <mailto:[email protected]>)
>>     > (gcc version 4.1.1 20070105 (Red Hat
>>     > 4.1.1-52)) #1
>>     >
>>     > Free PBX is the idea.
>>     >
>>     > Connected one IP phone and two Soft phones(One is 3CX
>>     > version
>>     > 4.0.10858.0 and other is X-lite version 3.0 build 29712 ).
>>     >
>>     > Given Nubering 400,401,402 call's is successful ,but while
>>     > doing
>>     > transfer ,line is going to hold but calls are not
>>     > connected  is no
>>     > tranfering happening,
>>     >
>>     > I need to know about call pickup,call FW and IVR detls
>>     > I searched so many sites, atlast i found that features.conf
>>     > doing all
>>     > these option...
>>     >
>>     > thanks and regards,
>>     >
>>     > Gireesh
>>     >
>>     > On 3/23/11, John Joseph <[email protected]
>>     <mailto:[email protected]>>
>>     > wrote:
>>     > > Hi Giresh
>>     > >    Can you please let us know which asterisk
>>     > are you using, If you are using
>>     > > AstersikNow or FreePBX or similar one; this option
>>     > could be configured ,
>>     > > through web interface.
>>     > >
>>     > > One way to get good answers or feedback is to post
>>     > your problems in details,
>>     > > so that others could really motivated to help you,
>>     > Please encourage others
>>     > > to help u
>>     > > Thanks
>>     > > Joseph John (Saji)
>>     > >
>>     > >
>>     > >
>>     > >
>>     > > --- On Wed, 23/3/11, gireesh thenayil rajappan nair
>>     > <[email protected] <mailto:[email protected]>>
>>     > > wrote:
>>     > >
>>     > > From: gireesh thenayil rajappan nair <[email protected]
>>     <mailto:[email protected]>>
>>     > > Subject: Re: [ILUG-Cochin.org] Asterisk features.conf
>>     > pls help
>>     > > To: "This List discusses GNU/Linux & GNU, GPL
>>     > Software"
>>     > > <[email protected]
>>     <mailto:[email protected]>>
>>     > > Date: Wednesday, 23 March, 2011, 1:20 PM
>>     > >
>>     > > Dear All,
>>     > >
>>     > > my Asterisk pbx the call forward/call transfer and
>>     > other facilities are not
>>     > > working ...
>>     > >
>>     > > pls help me to edit this in features.conf and
>>     > > features_general_additional.conf  
>>     > features_general_custom.conf
>>     > >
>>     > >
>>     > >
>>     > > regards,
>>     > >
>>     > > Gireesh
>>     > >
>>     > >
>>     > > -----Inline Attachment Follows-----
>>     > >
>>     > > _______________________________________________
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>>     > >
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