The problem is in your Asterisk, not in PJSIP. You should ask an Asterisk mailing lists for how to enable call transfer. Adrian On Mar 25, 2010, at 7:30 AM, Premalatha Kuppan wrote: > Can anyone please help me ? > > On Wed, Mar 24, 2010 at 7:17 PM, Premalatha Kuppan <premalatha at ngintech.com > > wrote: > Hi, > > Iam doing Call transfer using PJSIP. > > My scenario is like this, > > 1. B calls PJSIP(A) (B---->A) > 2. At A, iam getting press a to answer or h to hold, i gave teh > option 'x' to transfer the call and it promted for URL: gave the uri > (A-->C) > 3. REFER request is going, but after sending this i get 603 Declined > from the Asterisk server (I use asterisk server for my testing). > > Note: B & C: Xlite phone on linux machine; A -> PJSIP > > Anyone has any idea? > > > --end msg-- > 19:29:08.044 evsub0x18799c Subscription state changed NULL --> SENT > >>> 19:29:08.045 pjsua_core.c RX 525 bytes Response msg 603/ > REFER/cseq=21530 (rdata0x177e9c) from UDP 192.168.0.66:5060: > SIP/2.0 603 Declined > Via: SIP/2.0/UDP 192.168.0.162:5060;branch=z9hG4bKPjZ. > 2.kmMABJbELwWq1CuI8knPkJHOfgao;received=192.168.0.162;rport=5060 > From: <sip:700@192.168.0.162>;tag=trr9Zqw3lfv8PrEMgQFH7lCdfUZXP3ma > To: "Incoming" <sip:715 at 192.168.0.66>;tag=as0fbd005c > Call-ID: 7a66a2556acd9efe6ad5335c54860e2a at 192.168.0.66 > CSeq: 21530 REFER > Server: Asterisk PBX 1.6.1.6 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO > Supported: replaces, timer > Contact: <sip:715 at 192.168.0.66> > Content-Length: 0 > > > Please help. > > Thanks, > Premalatha > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100325/37a292b2/attachment.html>
