Call transfer

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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
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