Update Display information on Call Transfer

Asterisk Asterisk SIP
1

Hello,

I have set send_pai=yes, send_rpid=yes and connected_line_method = update in pjsip.conf.
This works fine on displaying the Name of the connected line, if I place a direct call to someone.

But if I make a transfer of the call, the Display Information get’s no Update.

Example Szenario:
Extension 1114 (Sal Smith) calls
Extension 1113 (Laverne Roberts)
1113 picks up the phone, and makes a call transfer to Extension 1110 (Hollis Justiy).
Extension 1110 Picks up the phone, talks to 1113, and afterwards 1113 is pressing the transfer button.
Now Extension 1110 and 1114 are connected.
The connected line on both extension now displays 1113 (Laverne Roberts). Why is the Display not updating in that situation?

Here is the asterisk output:

    -- Executing [1113@Long-Distance:1] NoOp("PJSIP/1114-00000024", "") in new stack
    -- Executing [1113@Long-Distance:2] Set("PJSIP/1114-00000024", "CDR_PROP(disable)=1") in new stack
    -- Executing [1113@Long-Distance:3] Goto("PJSIP/1114-00000024", "Internal-Main,1113,1") in new stack
    -- Goto (Internal-Main,1113,1)
    -- Executing [1113@Internal-Main:1] Verbose("PJSIP/1114-00000024", "1, "User 1114 dialed 1113."") in new stack
  "User 1114 dialed 1113."
    -- Executing [1113@Internal-Main:2] Set("PJSIP/1114-00000024", "SAC_DIALED_EXTEN=1113") in new stack
    -- Executing [1113@Internal-Main:3] GotoIf("PJSIP/1114-00000024", "0?dialed-BUSY,1:") in new stack
    -- Executing [1113@Internal-Main:4] Dial("PJSIP/1114-00000024", "PJSIP/1113,30") in new stack
    -- Called PJSIP/1113
    -- PJSIP/1113-00000025 is ringing
       > 0x7fbfb0029d00 -- Strict RTP learning after remote address set to: 192.168.150.62:11982
    -- PJSIP/1113-00000025 answered PJSIP/1114-00000024
       > 0x7fbfb0014ff0 -- Strict RTP learning after remote address set to: 192.168.150.57:12446
    -- Channel PJSIP/1113-00000025 joined 'simple_bridge' basic-bridge <a4fd1454-b852-4fba-aa96-86d450024c04>
    -- Channel PJSIP/1114-00000024 joined 'simple_bridge' basic-bridge <a4fd1454-b852-4fba-aa96-86d450024c04>
       > Bridge a4fd1454-b852-4fba-aa96-86d450024c04: switching from simple_bridge technology to native_rtp
       > Locally RTP bridged 'PJSIP/1114-00000024' and 'PJSIP/1113-00000025' in stack
       > 0x7fbfb0014ff0 -- Strict RTP switching to RTP target address 192.168.150.57:12446 as source
       > 0x7fbfb0029d00 -- Strict RTP switching to RTP target address 192.168.150.62:11982 as source
       > Bridge a4fd1454-b852-4fba-aa96-86d450024c04: switching from native_rtp technology to simple_bridge
    -- Started music on hold, class 'default', on channel 'PJSIP/1114-00000024'
       > 0x7fbfb0014ff0 -- Strict RTP learning complete - Locking on source address 192.168.150.57:12446
    -- Executing [1110@Long-Distance:1] NoOp("PJSIP/1113-00000026", "") in new stack
    -- Executing [1110@Long-Distance:2] Set("PJSIP/1113-00000026", "CDR_PROP(disable)=1") in new stack
    -- Executing [1110@Long-Distance:3] Goto("PJSIP/1113-00000026", "Internal-Main,1110,1") in new stack
    -- Goto (Internal-Main,1110,1)
    -- Executing [1110@Internal-Main:1] Verbose("PJSIP/1113-00000026", "1, "User 1113 dialed 1110."") in new stack
  "User 1113 dialed 1110."
    -- Executing [1110@Internal-Main:2] Set("PJSIP/1113-00000026", "SAC_DIALED_EXTEN=1110") in new stack
    -- Executing [1110@Internal-Main:3] GotoIf("PJSIP/1113-00000026", "0?dialed-BUSY,1:") in new stack
    -- Executing [1110@Internal-Main:4] Dial("PJSIP/1113-00000026", "PJSIP/1110,30") in new stack
    -- Called PJSIP/1110
    -- PJSIP/1110-00000027 is ringing
       > 0x7fbf6010ea90 -- Strict RTP learning after remote address set to: 192.168.150.72:12442
    -- PJSIP/1110-00000027 answered PJSIP/1113-00000026
       > 0x7fbf600fe330 -- Strict RTP learning after remote address set to: 192.168.150.62:11984
    -- Channel PJSIP/1110-00000027 joined 'simple_bridge' basic-bridge <623fb81d-8221-403b-ad98-a500ea6d7e86>
    -- Channel PJSIP/1113-00000026 joined 'simple_bridge' basic-bridge <623fb81d-8221-403b-ad98-a500ea6d7e86>
       > Bridge 623fb81d-8221-403b-ad98-a500ea6d7e86: switching from simple_bridge technology to native_rtp
       > Locally RTP bridged 'PJSIP/1113-00000026' and 'PJSIP/1110-00000027' in stack
       > 0x7fbf600fe330 -- Strict RTP switching to RTP target address 192.168.150.62:11984 as source
       > 0x7fbf6010ea90 -- Strict RTP switching to RTP target address 192.168.150.72:12442 as source
    -- Channel PJSIP/1114-00000024 left 'simple_bridge' basic-bridge <a4fd1454-b852-4fba-aa96-86d450024c04>
    -- Stopped music on hold on PJSIP/1114-00000024
    -- Channel PJSIP/1113-00000026 left 'native_rtp' basic-bridge <623fb81d-8221-403b-ad98-a500ea6d7e86>
    -- Channel PJSIP/1114-00000024 swapped with PJSIP/1113-00000026 into 'native_rtp' basic-bridge <623fb81d-8221-403b-ad98-a500ea6d7e86>
       > Bridge 623fb81d-8221-403b-ad98-a500ea6d7e86: switching from native_rtp technology to simple_bridge
    -- Channel PJSIP/1113-00000025 left 'simple_bridge' basic-bridge <a4fd1454-b852-4fba-aa96-86d450024c04>
  == Spawn extension (Internal-Main, 1110, 4) exited non-zero on 'PJSIP/1113-00000026'
    -- Executing [h@Internal-Main:1] Hangup("PJSIP/1113-00000026", "") in new stack
  == Spawn extension (Internal-Main, h, 1) exited non-zero on 'PJSIP/1113-00000026'
       > 0x7fbf6010ea90 -- Strict RTP learning complete - Locking on source address 192.168.150.72:12442

Thanks!
Best regards
Manuel

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