I can make call transfer from Xlite phones. Here, 1. C(xlitephone - 602) calls B(xlitephone - 601) 2. B transfer to our PJSIP (600) This scenario is successful. But when i make call transfer using our PJSIP, it fails. Am i missing anything ? Please help. *Log: * ** From: "Incoming" <sip:601@192.168.0.66 <sip%3A601 at 192.168.0.66>> To: <sip:600 at 192.168.0.162 <sip%3A600 at 192.168.0.162>> Press a to answer or h to reject call x Transfering current call [0] "Incoming" <sip:601 at 192.168.0.66<sip%3A601 at 192.168.0.66> > Buddy list: -none- Choices: 0 For current dialog. -1 All 0 buddies in buddy list [1 - 0] Select from buddy list URL An URL <Enter> Empty input (or 'q') to cancel *Transfer to URL: sip:602 at 192.168.0.66 <sip%3A602 at 192.168.0.66>* 17:09:55.211 pjsua_core.c TX 630 bytes Request msg REFER/cseq=6173 (tdta0x187920) to UDP 192.168.0.66:5060: *REFER sip:601 at 192.168.0.66 <sip%3A601 at 192.168.0.66> SIP/2.0* Via: SIP/2.0/UDP 192.168.0.162:5060 ;rport;branch=z9hG4bKPjetXVACTCGOQ5-3cpIzEdzKu-tpevO0No Max-Forwards: 70 From: <sip:600@192.168.0.162 <sip%3A600 at 192.168.0.162> >;tag=v1VZeMpIB5UzciOLqyCVX-Kv4a6qHCrG To: "Incoming" <sip:601 at 192.168.0.66 <sip%3A601 at 192.168.0.66> >;tag=as68b4c293 Contact: <sip:192.168.0.162:5060> Call-ID: 2fd7f9d85bb83f613f3e2f0900e3d897 at 192.168.0.66 CSeq: 6173 REFER Event: refer Expires: 600 Accept: message/sipfrag;version=2.0 Allow-Events: presence, message-summary, refer *Refer-To: sip:602 at 192.168.0.66 <sip%3A602 at 192.168.0.66>* *Referred-By: <sip:600 at 192.168.0.162 <sip%3A600 at 192.168.0.162>>* User-Agent: PJSUA v1.5.5/arm-unknown-linux-gnu Content-Length: 0 --end msg-- 17:09:55.211 evsub0x17a394 Subscription state changed NULL --> SENT >>> 17:09:55.211 pjsua_core.c RX 524 bytes Response msg 603/REFER/cseq=6173 (rdata0x177d94) from UDP 192.168.0.66:5060: *SIP/2.0 603 Declined* Via: SIP/2.0/UDP 192.168.0.162:5060 ;branch=z9hG4bKPjetXVACTCGOQ5-3cpIzEdzKu-tpevO0No;received=192.168.0.162;rport=5060 From: <sip:600@192.168.0.162 <sip%3A600 at 192.168.0.162> >;tag=v1VZeMpIB5UzciOLqyCVX-Kv4a6qHCrG To: "Incoming" <sip:601 at 192.168.0.66 <sip%3A601 at 192.168.0.66> >;tag=as68b4c293 Call-ID: 2fd7f9d85bb83f613f3e2f0900e3d897 at 192.168.0.66 CSeq: 6173 REFER Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:601 at 192.168.0.66 <sip%3A601 at 192.168.0.66>> Content-Length: 0 --end msg-- 17:09:55.211 evsub0x17a394 Subscription state changed SENT --> TERMINATED 17:09:55.211 pjsua_call.c Xfer client subscription terminated 17:09:55.211 pjsua_call.c Warning: received NOTIFY without message body 17:10:00.221 evsub0x17a394 Subscription destroyed I appreciate your valuable input and time. Thank you so much. On Thu, Mar 25, 2010 at 3:21 PM, Adrian Georgescu <ag at ag-projects.com>wrote: > 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 <sip%3A700 at 192.168.0.162> >> >;tag=trr9Zqw3lfv8PrEMgQFH7lCdfUZXP3ma >> To: "Incoming" <sip:715 at 192.168.0.66 <sip%3A715 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 <sip%3A715 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 > > > > _______________________________________________ > 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... 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