[Call-Forwarding] What's the problem?

violetmch · February 18, 2019, 9:00am

Hi,
My system configuration consists of IPECS (PBX) in front of Asterisk.
I registered with IPECS through PJSIP and am using it.

Can anyone help me call Forwarding?

At the bottom is my setting.

Asterisk : 192.168.169.201
IPECS pbx : 192.168.168.2

[outbound-local]
exten = _9010.,1,Dial(PJSIP/${EXTEN}@2544_endpoint)
same = n,Hangup()
exten = _010.,1,Dial(PJSIP/${EXTEN}@2544_endpoint)
same = n,Hangup()
exten = _XXXX,1,Dial(PJSIP/${EXTEN}@2544_endpoint)
same = n,Hangup()

[from-external]
exten = 2536,1,NoOp(Incoming call!)
same = n,GoTo(outbound-local,2500,1)
same = n,Hangup()

<--- Received SIP request (935 bytes) from UDP:192.168.169.210:5060 --->
INVITE sip:2536@192.168.169.201 SIP/2.0
From: <sip:010XXXXXXXX@192.168.169.210>;tag=be9ae3f0-d2a9a8c0-13c4-55013-6446aa-28af2072-6446aa
To: <sip:2536@192.168.169.201>
Call-ID: b97f2398-d2a9a8c0-13c4-55013-6446aa-3a040d94-6446aa
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.169.210:5060;rport;branch=z9hG4bK-6446aa-87b40a7e-5982cbdc
Max-Forwards: 70
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,REFER,SUBSCRIBE,NOTIFY,MESSAGE,INFO,PRACK,UPDATE
P-Asserted-Identity: <sip:010XXXXXXXX@192.168.169.210:5060>
Supported: replaces
User-Agent: Ericsson-LG iPECS UCM v-Tmp1809-02.0.020
Contact: <sip:010XXXXXXXX@192.168.169.210:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 235

  == Setting global variable 'SIPDOMAIN' to '192.168.169.201'
<--- Transmitting SIP response (393 bytes) to UDP:192.168.169.210:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.169.210:5060;rport=5060;received=192.168.169.210;branch=z9hG4bK-6446aa-87b40a7e-5982cbdc
Call-ID: b97f2398-d2a9a8c0-13c4-55013-6446aa-3a040d94-6446aa
From: <sip:010XXXXXXXX@192.168.169.210>;tag=be9ae3f0-d2a9a8c0-13c4-55013-6446aa-28af2072-6446aa
To: <sip:2536@192.168.169.201>
CSeq: 1 INVITE
Server: Asterisk PBX 16.0.0
Content-Length:  0

    -- Executing [2536@from-external:1] NoOp("PJSIP/2539_endpoint-0000006b", "Incoming call!") in new stack
    -- Executing [2536@from-external:2] Goto("PJSIP/2539_endpoint-0000006b", "outbound-local,2500,1") in new stack
    -- Goto (outbound-local,2500,1)
    -- Executing [2500@outbound-local:1] Dial("PJSIP/2539_endpoint-0000006b", "PJSIP/2500@2544_endpoint") in new stack
    -- Called PJSIP/2500@2544_endpoint
<--- Transmitting SIP request (859 bytes) to UDP:192.168.168.210:5060 --->
INVITE sip:2500@192.168.168.210:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.169.201:5060;rport;branch=z9hG4bKPj196a911c-eb07-4b0c-9e0a-6b35088bf804
From: <sip:010XXXXXXXX@192.168.169.201>;tag=3269e472-60f3-4f6a-af04-b76e1bdb9c5b
To: <sip:2500@192.168.168.210>
Contact: <sip:asterisk@192.168.169.201:5060>
Call-ID: 8351487d-202c-4765-9701-bc312d2920a6
CSeq: 21001 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.0.0
Content-Type: application/sdp
Content-Length:   187

<--- Transmitting SIP request (859 bytes) to UDP:192.168.168.210:5060 --->
INVITE sip:2500@192.168.168.210:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.169.201:5060;rport;branch=z9hG4bKPj196a911c-eb07-4b0c-9e0a-6b35088bf804
From: <sip:010XXXXXXXX@192.168.169.201>;tag=3269e472-60f3-4f6a-af04-b76e1bdb9c5b
To: <sip:2500@192.168.168.210>
Contact: <sip:asterisk@192.168.169.201:5060>
Call-ID: 8351487d-202c-4765-9701-bc312d2920a6
CSeq: 21001 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.0.0
Content-Type: application/sdp
Content-Length:   187

<--- Transmitting SIP request (859 bytes) to UDP:192.168.168.210:5060 --->
INVITE sip:2500@192.168.168.210:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.169.201:5060;rport;branch=z9hG4bKPj196a911c-eb07-4b0c-9e0a-6b35088bf804
From: <sip:010XXXXXXXX@192.168.169.201>;tag=3269e472-60f3-4f6a-af04-b76e1bdb9c5b
To: <sip:2500@192.168.168.210>
Contact: <sip:asterisk@192.168.169.201:5060>
Call-ID: 8351487d-202c-4765-9701-bc312d2920a6
CSeq: 21001 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.0.0
Content-Type: application/sdp
Content-Length:   187

<--- Transmitting SIP request (859 bytes) to UDP:192.168.168.210:5060 --->
INVITE sip:2500@192.168.168.210:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.169.201:5060;rport;branch=z9hG4bKPj196a911c-eb07-4b0c-9e0a-6b35088bf804
From: <sip:010XXXXXXXX@192.168.169.201>;tag=3269e472-60f3-4f6a-af04-b76e1bdb9c5b
To: <sip:2500@192.168.168.210>
Contact: <sip:asterisk@192.168.169.201:5060>
Call-ID: 8351487d-202c-4765-9701-bc312d2920a6
CSeq: 21001 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.0.0
Content-Type: application/sdp
Content-Length:   187

<--- Received SIP request (460 bytes) from UDP:192.168.169.210:5060 --->
CANCEL sip:2536@192.168.169.201 SIP/2.0
From: <sip:010XXXXXXXX@192.168.169.210>;tag=be9ae3f0-d2a9a8c0-13c4-55013-6446aa-28af2072-6446aa
To: <sip:2536@192.168.169.201>
Call-ID: b97f2398-d2a9a8c0-13c4-55013-6446aa-3a040d94-6446aa
CSeq: 1 CANCEL
Via: SIP/2.0/UDP 192.168.169.210:5060;rport;branch=z9hG4bK-6446aa-87b40a7e-5982cbdc
Max-Forwards: 70
Supported: 100rel,timer,replaces
User-Agent: Ericsson-LG iPECS UCM v-Tmp1809-02.0.020
Content-Length: 0


<--- Transmitting SIP response (430 bytes) to UDP:192.168.169.210:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.169.210:5060;rport=5060;received=192.168.169.210;branch=z9hG4bK-6446aa-87b40a7e-5982cbdc
Call-ID: b97f2398-d2a9a8c0-13c4-55013-6446aa-3a040d94-6446aa
From: <sip:010XXXXXXXX@192.168.169.210>;tag=be9ae3f0-d2a9a8c0-13c4-55013-6446aa-28af2072-6446aa
To: <sip:2536@192.168.169.201>;tag=7bab22cc-2adb-44a8-8692-48c5115a83bc
CSeq: 1 CANCEL
Server: Asterisk PBX 16.0.0
Content-Length:  0


<--- Transmitting SIP response (446 bytes) to UDP:192.168.169.210:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.169.210:5060;rport=5060;received=192.168.169.210;branch=z9hG4bK-6446aa-87b40a7e-5982cbdc
Call-ID: b97f2398-d2a9a8c0-13c4-55013-6446aa-3a040d94-6446aa
From: <sip:010XXXXXXXX@192.168.169.210>;tag=be9ae3f0-d2a9a8c0-13c4-55013-6446aa-28af2072-6446aa
To: <sip:2536@192.168.169.201>;tag=7bab22cc-2adb-44a8-8692-48c5115a83bc
CSeq: 1 INVITE
Server: Asterisk PBX 16.0.0
Content-Length:  0


  == Spawn extension (outbound-local, 2500, 1) exited non-zero on 'PJSIP/2539_endpoint-0000006b'
[Feb 18 17:55:16] ERROR[18701]: cdr_odbc.c:174 odbc_log: Unable to retrieve database handle.  CDR failed.
<--- Received SIP request (524 bytes) from UDP:192.168.169.210:5060 --->
ACK sip:2536@192.168.169.201 SIP/2.0
From: <sip:010XXXXXXXX@192.168.169.210>;tag=be9ae3f0-d2a9a8c0-13c4-55013-6446aa-28af2072-6446aa
To: <sip:2536@192.168.169.201>;tag=7bab22cc-2adb-44a8-8692-48c5115a83bc
Call-ID: b97f2398-d2a9a8c0-13c4-55013-6446aa-3a040d94-6446aa
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.169.210:5060;rport;branch=z9hG4bK-6446aa-87b40a7e-5982cbdc
Max-Forwards: 70
User-Agent: Ericsson-LG iPECS UCM v-Tmp1809-02.0.020
Contact: <sip:010XXXXXXXX@192.168.169.210:5060;transport=udp>
Content-Length: 0


<--- Transmitting SIP request (859 bytes) to UDP:192.168.168.210:5060 --->
INVITE sip:2500@192.168.168.210:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.169.201:5060;rport;branch=z9hG4bKPj196a911c-eb07-4b0c-9e0a-6b35088bf804
From: <sip:010XXXXXXXX@192.168.169.201>;tag=3269e472-60f3-4f6a-af04-b76e1bdb9c5b
To: <sip:2500@192.168.168.210>
Contact: <sip:asterisk@192.168.169.201:5060>
Call-ID: 8351487d-202c-4765-9701-bc312d2920a6
CSeq: 21001 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.0.0
Content-Type: application/sdp
Content-Length:   187

<--- Transmitting SIP request (859 bytes) to UDP:192.168.168.210:5060 --->
INVITE sip:2500@192.168.168.210:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.169.201:5060;rport;branch=z9hG4bKPj196a911c-eb07-4b0c-9e0a-6b35088bf804
From: <sip:010XXXXXXXX@192.168.169.201>;tag=3269e472-60f3-4f6a-af04-b76e1bdb9c5b
To: <sip:2500@192.168.168.210>
Contact: <sip:asterisk@192.168.169.201:5060>
Call-ID: 8351487d-202c-4765-9701-bc312d2920a6
CSeq: 21001 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.0.0
Content-Type: application/sdp
Content-Length:   187

david551 · February 18, 2019, 9:44am

The B side INVITE is not reaching the peer or the response from the B side peer is not reaching Asterisk.

In future, please use </> to mark dialplans as pre-formatted text, on the forum.

violetmch · February 19, 2019, 1:59am

Thanks to you, it’s been solved.

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