Hi Kurian,

Is you worked in asterisk before ,can you help me ...


Thanks ans regards,

Gireesh

On Fri, Mar 25, 2011 at 5:52 PM, Kurian Thayil <[email protected]>wrote:

>  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
>
> 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]> 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]> wrote:
>>
>> > From: gireesh thenayil rajappan nair <[email protected]>
>> > Subject: Re: [ILUG-Cochin.org] Asterisk features.conf pls help
>> > To: "This List discusses GNU/Linux & GNU, GPL Software" <
>> [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])
>> > (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]>
>> > 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]>
>> > > wrote:
>> > >
>> > > From: gireesh thenayil rajappan nair <[email protected]>
>> > > Subject: Re: [ILUG-Cochin.org] Asterisk features.conf
>> > pls help
>> > > To: "This List discusses GNU/Linux & GNU, GPL
>> > Software"
>> > > <[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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