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----- > > > > > > _______________________________________________ > > > Indian Libre User Group Cochin Mailing List > > > http://www.ilug-cochin.org/mailing-list/ > > > > http://mail.ilug-cochin.org/mailman/listinfo/mailinglist_ilug-cochin.org > > > #[email protected] <mailto:[email protected]> > > > > > > > > > > > > _______________________________________________ > > > Indian Libre User Group Cochin Mailing List > > > http://www.ilug-cochin.org/mailing-list/ > > > > http://mail.ilug-cochin.org/mailman/listinfo/mailinglist_ilug-cochin.org > > > #[email protected] <mailto:[email protected]> > > > > > > > _______________________________________________ > > Indian Libre User Group Cochin Mailing List > > http://www.ilug-cochin.org/mailing-list/ > > > http://mail.ilug-cochin.org/mailman/listinfo/mailinglist_ilug-cochin.org > > #[email protected] <mailto:[email protected]> > > > > > > _______________________________________________ > Indian Libre User Group Cochin Mailing List > http://www.ilug-cochin.org/mailing-list/ > http://mail.ilug-cochin.org/mailman/listinfo/mailinglist_ilug-cochin.org > #[email protected] <mailto:[email protected]> > > > > _______________________________________________ > Indian Libre User Group Cochin Mailing List > http://www.ilug-cochin.org/mailing-list/ > http://mail.ilug-cochin.org/mailman/listinfo/mailinglist_ilug-cochin.org > #[email protected]
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