Haven't you read my email? 1. Wrong list 2. Missing log entries (set debug 4, set verbose 4)
klaus Max Alex schrieb: > Hi All, > Thanks for your reply. > I got this refer message in asterisk. > but there is not any active channel of blind transfer. > ---------------------- > REFER sip:[email protected] <mailto:sip%[email protected]> SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0 > From: <sip:[email protected]:5060;transport=udp>;tag=3699e1bcbed17687 > To: "1101" <sip:[email protected] > <mailto:sip%[email protected]>>;tag=as32ed6c48 > Contact: <sip:[email protected]:5060;transport=udp> > Supported: replaces, path > Refer-To: <sip:[email protected] > <mailto:sip%[email protected]>> > Referred-By: <sip:[email protected] <mailto:sip%[email protected]>> > Call-ID: [email protected] > <mailto:[email protected]> > CSeq: 34526 REFER > User-Agent: Grandstream BT200 1.1.6.46 > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > Content-Length: 0 > > <-------------> > --- (14 headers 0 lines) --- > Call [email protected] > <mailto:[email protected]> got a SIP call > transfer from caller: (REFER)! > SIP transfer to extension 1631xxxx...@outgoing by [email protected] > <mailto:[email protected]> > localhost*CLI> > <--- Transmitting (NAT) to 192.168.1.30:5060 <http://192.168.1.30:5060> ---> > SIP/2.0 202 Accepted > Via: SIP/2.0/UDP > 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0;received=192.168.1.30 > From: <sip:[email protected]:5060;transport=udp>;tag=3699e1bcbed17687 > To: "1101" <sip:[email protected] > <mailto:sip%[email protected]>>;tag=as32ed6c48 > Call-ID: [email protected] > <mailto:[email protected]> > CSeq: 34526 REFER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected] <mailto:sip%[email protected]>> > Content-Length: 0 > > > <------------> > ---------------------------------------- > Is there any options we need to enable in asterisk or grandstream phone? > I have already user transfer option 'Tt' in dialplan of this. > Please provide me some help. > Thanks in advance!! > > Thanks, > Max Alex > Voip Developer > > > > On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion > <[email protected] <mailto:[email protected]>> wrote: > > Max Alex wrote: > > Hi All, > > I have working asterisk version 1.4.24. > > I have a blind transfer issue with grandstream bt200. > > Does it work with other phones? That means is it a Grandstream isue or a > general issue? > > > I have updated the latest firmware to the phone. > > The phone is sending the *refer* to asterisk but asterisk is not > able to > > connect with the another call > > Why? some log messages would help us helping you. > > > that i have checked in sip debug. > > I am using transfer button of the grandstream phone. > > Can anybody provide help for this issue? > > Please ask again on the user mailing lists and provide some log messages > > > Thanks in advance!! > > > > Thanks, > > Max Alex > > Voip Developer > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
