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] <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] <sip%[email protected]>>;tag=as32ed6c48 Contact: <sip:[email protected]:5060;transport=udp> Supported: replaces, path Refer-To: <sip:[email protected] <sip%[email protected]>> Referred-By: <sip:[email protected] <sip%[email protected]>> Call-ID: [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] got a SIP call transfer from caller: (REFER)! SIP transfer to extension 1631xxxx...@outgoing by [email protected] localhost*CLI> <--- Transmitting (NAT) to 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] <sip%[email protected]>>;tag=as32ed6c48 Call-ID: [email protected] CSeq: 34526 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected] <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] > 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 >
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