Caller ID not diaplyed on incoming DID call

voip.linux

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Aug 26, 2008
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#1
Helo all,
I have setup SIP trunk and incoming call to DID connecting with Extension via " InBound ROute"

However, I can't see callerid on my IPPhone while incoming call!!!
Logs show that it was detected correctly : CallerID is "anonymous" <7141111111>"
But it dials DID from Unknown instead of using this 7141111111:
From: "Unknown" <sip:Unknown@ELASTIX_SRV_IP>;tag=as65548cde

How to enable caller_ID display ?


LOG flow:


Executing [8008011111@from-trunk:1] Set("SIP/192.168.0.1-09f16a70", "__FROM_DID=8008011111") in new stack
-- Executing [8008011111@from-trunk:2] GotoIf("SIP/192.168.0.1-09f16a70", "1 ?cidok") in new stack
-- Goto (from-trunk,8008011111,4)
-- Executing [8008011111@from-trunk:4] NoOp("SIP/192.168.0.1-09f16a70", "CallerID is "anonymous" <7141111111>") in new stack
-- Executing [8008011111@from-trunk:8] Goto("SIP/192.168.0.1-09f16a70", "from-did-direct|876|1") in new stack
-- Goto (from-did-direct,876,1)
-- Executing [876@from-did-direct:1] Macro("SIP/192.168.0.1-09f16a70", "exten-vm|876|876") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/192.168.0.1-09f16a70", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/192.168.0.1-09f16a70", "user-callerid: anonymous 7141111111") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/192.168.0.1-09f16a70", "AMPUSER=7141111111") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/192.168.0.1-09f16a70", "REALCALLERIDNUM=7141111111") in new stack
-- Executing [s@macro-user-callerid:6] NoOp("SIP/192.168.0.1-09f16a70", "REALCALLERIDNUM is 7141111111") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/192.168.0.1-09f16a70", "1?continue") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/192.168.0.1-09f16a70", "Using CallerID "anonymous" <7141111111>") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/192.168.0.1-09f16a70", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-dial:3] AGI("SIP/192.168.0.1-09f16a70", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'anonymous' number is '7141111111'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 876 to extension map
-- dialparties.agi: Extension 876 cf is disabled
-- dialparties.agi: Extension 876 do not disturb is disabled
dialparties.agi: Extension 876 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 876
-- dialparties.agi: dbset CALLTRACE/876 to 7141111111
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:10] Dial("SIP/192.168.0.1-09f16a70", "SIP/876|15|tr") in new stack
Audio is at ELASTIX_SRV_IP port 11822
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.12:5060:
INVITE sip:876@192.168.0.12:5060;rinstance=4fadfb23cf7b6369 SIP/2.0
Via: SIP/2.0/UDP ELASTIX_SRV_IP:5060;branch=z9hG4bK753cdbf0;rport
From: "Unknown" <sip:Unknown@ELASTIX_SRV_IP>;tag=as65548cde
To: <sip:876@192.168.0.12:5060;rinstance=4fadfb23cf7b6369>
Contact: <sip:Unknown@ELASTIX_SRV_IP>
 

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