This is what is actually going on: Call is made to test-peer from number 123456789
SIP/2.0 180 Ringing Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4baae0d7;rport From: "Empty" <sip:[email protected]>;tag=as24ef1afd To: "Test Peer" <sip:[email protected]>;tag=93AFFFD9-7DF89662 CSeq: 102 INVITE Call-ID: [email protected]:5060 Contact: <sip:[email protected]> User-Agent: PolycomSoundPointIP-SPIP_650-UA/4.0.11.0583 Allow-Events: conference,talk,hold Accept-Language: en Content-Length: 0 Polycom redirects it to number 9999 SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4baae0d7;rport From: "Empty" <sip:[email protected]>;tag=as24ef1afd To: "Test Peer" <sip:[email protected]>;tag=93AFFFD9-7DF89662 CSeq: 102 INVITE Call-ID: [email protected]:5060 Contact: <sip:[email protected];user=phone> User-Agent: PolycomSoundPointIP-SPIP_650-UA/4.0.11.0583 Accept-Language: en Diversion: "Test Peer" <sip:[email protected]>;reason=deflection Content-Length: 0 I would like that the peer at number 9999 is receiving the real number 123456789, but it is receiving test-peer internal number. вт, 25 июн. 2019 г. в 18:05, Doug Lytle <[email protected]>: > >>> Surely that is "call forwarding", which is quite different from either > a blind or attended transfer? > > That would be correct. > > The forward button on the polycom phones just do a redirect to the > destination extension or external phone number. > > Doug > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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