SIP Trunk

Discussion in 'General' started by tsku, Oct 22, 2010.

  1. tsku

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    Hi all,

    I'm trying to configure Elastix 2.0 to have a SIP trunk for call termination to VoIP Clearing House but no luck to make it work. When try to pass call to clearing house, it is always show circuit busy. However, IAX trunk is working fine for me.

    I had used SJPHONE to try the SIP account and it work fine for me, mean the login information and remote server are working fine.

    "sip show peers" show as below, looks like my Elastix is not able to connect to remote server. Any idea to make it work? Thanks in advance.


    Name/username Host Dyn Nat ACL Port Status
    2231120/2231120 202.211.228.41 5060 Unmonitored
     
  2. Lee Sharp

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    We can just guess at stuff...

    Or you can type "asterisk -r" into a shell and give us the output when you fail to make a call.
     
  3. tsku

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    Hi Lee,

    verbosity is at least 3
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    -- Executing [990162166917@from-internal:1] Macro("SIP/1231805-0000028a", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/1231805-0000028a", "AMPUSER=1231805") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1231805-0000028a", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1231805-0000028a", "1?Set(REALCALLERIDNUM=1231805)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/1231805-0000028a", "AMPUSER=1231805") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/1231805-0000028a", "AMPUSERCIDNAME=508") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1231805-0000028a", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/1231805-0000028a", "AMPUSERCID=1231805") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/1231805-0000028a", "CALLERID(all)="508" <1231805>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1231805-0000028a", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1231805-0000028a", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/1231805-0000028a", "Using CallerID "508" <1231805>") in new stack
    -- Executing [990162166917@from-internal:2] Set("SIP/1231805-0000028a", "_NODEST=") in new stack
    -- Executing [990162166917@from-internal:3] Macro("SIP/1231805-0000028a", "record-enable,1231805,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1231805-0000028a", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/1231805-0000028a", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/1231805-0000028a", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/1231805-0000028a", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/1231805-0000028a", "1?MacroExit()") in new stack
    -- Executing [990162166917@from-internal:4] Macro("SIP/1231805-0000028a", "dialout-trunk,4,0162166917,,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/1231805-0000028a", "DIAL_TRUNK=4") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1231805-0000028a", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1231805-0000028a", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/1231805-0000028a", "DIAL_NUMBER=0162166917") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/1231805-0000028a", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/1231805-0000028a", "OUTBOUND_GROUP=OUT_4") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1231805-0000028a", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1231805-0000028a", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/1231805-0000028a", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1231805-0000028a", "outbound-callerid,4") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1231805-0000028a", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1231805-0000028a", "0?Set(REALCALLERIDNUM=1231805)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1231805-0000028a", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/1231805-0000028a", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/1231805-0000028a", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/1231805-0000028a", "TRUNKOUTCID=oce-sip-outgoing-redtone") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1231805-0000028a", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1231805-0000028a", "1?Set(CALLERID(all)=oce-sip-outgoing-redtone)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1231805-0000028a", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1231805-0000028a", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1231805-0000028a", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1231805-0000028a", "0?AGI(fixlocalprefix)") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/1231805-0000028a", "OUTNUM=0162166917") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/1231805-0000028a", "custom=SIP/2231120") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1231805-0000028a", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/1231805-0000028a", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1231805-0000028a", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1231805-0000028a", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1231805-0000028a", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1231805-0000028a", "SIP/2231120/0162166917,300,") in new stack
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    -- Called 2231120/0162166917
    -- SIP/2231120-0000028b is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/1231805-0000028a", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/1231805-0000028a", "s-CONGESTION,1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/1231805-0000028a", "RC=21") in new stack
    -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/1231805-0000028a", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/1231805-0000028a", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/1231805-0000028a", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/1231805-0000028a", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/1231805-0000028a", "CALLERID(number)=1231805") in new stack
    -- Executing [990162166917@from-internal:5] Macro("SIP/1231805-0000028a", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/1231805-0000028a", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/1231805-0000028a", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/1231805-0000028a", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/1231805-0000028a", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/1231805-0000028a> Playing 'all-circuits-busy-now.gsm' (language 'en')
    == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/1231805-0000028a' in macro 'outisbusy'
    == Spawn extension (from-internal, 990162166917, 5) exited non-zero on 'SIP/1231805-0000028a'
    -- Executing [h@from-internal:1] Macro("SIP/1231805-0000028a", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1231805-0000028a", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/1231805-0000028a", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1231805-0000028a", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1231805-0000028a", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1231805-0000028a", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,12)
    -- Executing [s@macro-hangupcall:12] Hangup("SIP/1231805-0000028a", "") in new stack
    == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/1231805-0000028a' in macro 'hangupcall'
    == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1231805-0000028a'
     
  4. Lee Sharp

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    I see the errors, but do not know why... Dicko? You got a clue?
     
  5. tsku

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    Hi Lee,

    I saw some threads said Elastix has some issue with SIP trunk but cant find solution on it. Is this true? Anyway, I will try to use asterisk to connect today to confirm whether problem happen to Elastix only.
     
  6. dicko

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    I would start by adding

    qualify=yes

    to his trunk called 2231120
     
  7. danardf

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    Hi all.

    That could be a nat issue too. But like Dicko said, first, put qualify=yes.
     
  8. tsku

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    Hi,

    Thanks for your advice. I tried with qualify=yes in my sip trunk and still cannot work.

    Without qualify=yes, "sip show peers" show sip trunk status unmonitored
    with qualify=yes, "sip show peers" show sip trunk status unreachable

    Below are the log for outgoing call.



    -- Executing [990162166917@from-internal:1] Macro("SIP/1231205-0000048c", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/1231205-0000048c", "AMPUSER=1231205") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1231205-0000048c", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1231205-0000048c", "1?Set(REALCALLERIDNUM=1231205)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/1231205-0000048c", "AMPUSER=1231205") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/1231205-0000048c", "AMPUSERCIDNAME=CK") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1231205-0000048c", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/1231205-0000048c", "AMPUSERCID=1231205") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/1231205-0000048c", "CALLERID(all)="CK" <1231205>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1231205-0000048c", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1231205-0000048c", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/1231205-0000048c", "Using CallerID "CK" <1231205>") in new stack
    -- Executing [990162166917@from-internal:2] Set("SIP/1231205-0000048c", "_NODEST=") in new stack
    -- Executing [990162166917@from-internal:3] Macro("SIP/1231205-0000048c", "record-enable,1231205,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1231205-0000048c", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/1231205-0000048c", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/1231205-0000048c", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/1231205-0000048c", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/1231205-0000048c", "1?MacroExit()") in new stack
    -- Executing [990162166917@from-internal:4] Macro("SIP/1231205-0000048c", "dialout-trunk,4,0162166917,,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/1231205-0000048c", "DIAL_TRUNK=4") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1231205-0000048c", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1231205-0000048c", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/1231205-0000048c", "DIAL_NUMBER=0162166917") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/1231205-0000048c", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/1231205-0000048c", "OUTBOUND_GROUP=OUT_4") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1231205-0000048c", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1231205-0000048c", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/1231205-0000048c", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1231205-0000048c", "outbound-callerid,4") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1231205-0000048c", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1231205-0000048c", "0?Set(REALCALLERIDNUM=1231205)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1231205-0000048c", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/1231205-0000048c", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/1231205-0000048c", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/1231205-0000048c", "TRUNKOUTCID=oce-sip-outgoing-redtone") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1231205-0000048c", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1231205-0000048c", "1?Set(CALLERID(all)=oce-sip-outgoing-redtone)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1231205-0000048c", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1231205-0000048c", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1231205-0000048c", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1231205-0000048c", "0?AGI(fixlocalprefix)") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/1231205-0000048c", "OUTNUM=0162166917") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/1231205-0000048c", "custom=SIP/2231120") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1231205-0000048c", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/1231205-0000048c", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1231205-0000048c", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1231205-0000048c", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1231205-0000048c", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1231205-0000048c", "SIP/2231120/0162166917,300,") in new stack
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/1231205-0000048c", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/1231205-0000048c", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/1231205-0000048c", "RC=20") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/1231205-0000048c", "20,1") in new stack
    -- Goto (macro-dialout-trunk,20,1)
    -- Executing [20@macro-dialout-trunk:1] Goto("SIP/1231205-0000048c", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/1231205-0000048c", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/1231205-0000048c", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/1231205-0000048c", "CALLERID(number)=1231205") in new stack
    -- Executing [990162166917@from-internal:5] Macro("SIP/1231205-0000048c", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/1231205-0000048c", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/1231205-0000048c", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/1231205-0000048c", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/1231205-0000048c", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/1231205-0000048c> Playing 'all-circuits-busy-now.gsm' (language 'en')
    -- <SIP/1231205-0000048c> Playing 'pls-try-call-later.gsm' (language 'en')
     
  9. danardf

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    qualify=yes, show only the trunk status.

    Try to put every nat parameters. So into your trunk nat=yes, and try to find in this forum, how to set the sip_nat.conf file.

    Restart Asterisk and try to make a call.
     
  10. tsku

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    Dear danardf,

    I tried nat=yes before but it doesn't help. FYI, the server is on Public IP and no NAT involved.

    Regards
     
  11. dicko

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    Indeed Franck it only shows the status,

    so rasterisk -x 'sip show peers'

    will confirm that he is talking to it it will say ok if it does

    then sip debug ip 202.211.228.41

    to identify what is not working

    reagrds

    dicko
     
  12. tsku

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    Hi Dicko,

    SIP set debug ip 202.211.228.41

    Audio is at 202.73.9.180 port 13912
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 202.211.228.41:5060:
    INVITE sip:0162166917@sip2.abc.com SIP/2.0
    Via: SIP/2.0/UDP 202.73.9.180:5060;branch=z9hG4bK3244bf3a;rport
    Max-Forwards: 70
    From: "oce-sip-outgoing-abc" <sip:Unknown@202.73.9.180>;tag=as63e10240
    To: <sip:0162166917@sip2.abc.com>
    Contact: <sip:Unknown@202.73.9.180>
    Call-ID: 74bb70c35c8829ae0a30258f1bbae10a@202.73.9.180
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.6.2.10
    Date: Mon, 25 Oct 2010 06:14:35 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 260

    v=0
    o=root 2231120 2231120 IN IP4 202.73.9.180
    s=Asterisk PBX 1.6.2.10
    c=IN IP4 202.73.9.180
    t=0 0
    m=audio 13912 RTP/AVP 8 3 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv

    ---
    -- Called 2231120/0162166917

    <--- SIP read from UDP:202.211.228.41:5060 --->
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 202.211.228.41:5060;branch=z9hG4bK3244bf3a;rport=5060
    From: "oce-sip-outgoing-abc" <sip:Unknown@202.73.9.180>;tag=as63e10240
    To: <sip:0162166917@sip2.abc.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b7c9
    Call-ID: 74bb70c35c8829ae0a30258f1bbae10a@202.73.9.180
    CSeq: 102 INVITE
    Server: Sip EXpress router (0.9.6 (i386/linux))
    Content-Length: 0
    Warning: 392 202.211.228.41:5060 "Noisy feedback tells: pid=2447 req_src_ip=202.73.9.180 req_src_port=5060 in_uri=sip:0162166917@sip2.abc.com out_uri=sip:0162166917@sip2.abc.com via_cnt==1"


    <------------->
    --- (9 headers 0 lines) ---
    Transmitting (NAT) to 202.211.228.41:5060:
    ACK sip:0162166917@sip2.abc.com SIP/2.0
    Via: SIP/2.0/UDP 202.73.9.180:5060;branch=z9hG4bK3244bf3a;rport
    Max-Forwards: 70
    From: "oce-sip-outgoing-abc" <sip:Unknown@202.73.9.180>;tag=as63e10240
    To: <sip:0162166917@sip2.abc.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b7c9
    Contact: <sip:Unknown@202.73.9.180>
    Call-ID: 74bb70c35c8829ae0a30258f1bbae10a@202.73.9.180
    CSeq: 102 ACK
    User-Agent: Asterisk PBX 1.6.2.10
    Content-Length: 0
     
  13. dicko

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    I don't think you have enough sip traffic there to be very usuful but it looks like your authorities don't agree.

    you might want to try

    insecure=very

    in the trunk settings
     
  14. tsku

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    Hi Dicko,

    I'm seeing message below and from is always show unknown, is this the root cause why I can't pass call to clearing house because from is unknown?

    Retransmitting #2 (NAT) to 202.211.228.41:5060:
    OPTIONS sip:sip2.abc.com SIP/2.0
    Via: SIP/2.0/UDP 202.73.9.180:5060;branch=z9hG4bK06bd20f8;rport
    Max-Forwards: 70
    From: "Unknown" <sip:Unknown@202.73.9.180>;tag=as6baa4c5f
    To: <sip:sip2.abc.com>
    Contact: <sip:Unknown@202.73.9.180>
    Call-ID: 0730d30319f6a2bf7f7332d10b2daae2@202.73.9.180
    CSeq: 102 OPTIONS
    User-Agent: Asterisk PBX 1.6.2.10
    Date: Mon, 25 Oct 2010 08:28:32 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Length: 0
     
  15. danardf

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    Don't forget in your router, to forward every SIP and RTP ports to Elastix server.
    5060 UDP and 10000 to 20000 UDP.
     
  16. tsku

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    Hi Danardf,

    This server is in DMZ with its own Public IP address and no NAT involve, firewall allow all traffic origin from this server and I don't think firewall or router will caused this issue.
     
  17. danardf

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    And why not? And if it was the case?

    Anyway, show me your config trunk. I can check it if you could have any problem.
     
  18. tsku

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    [2231120]
    disallow=all
    username=2231120
    type=peer
    secret=xjnjnvknliok
    host=202.211.228.41
    allow=alaw
    allow=gsm
    context=from-trunk-sip-2231120

    I tried to stop the firewall rules but still not working and it proved that firewall is not the root caused for this,
     
  19. danardf

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    Try to set this config like that:

    disallow=all
    allow=alaw
    allow=gsm
    username=2231120
    type=peer
    secret=xjnjnvknliok
    host=202.211.228.41
    context=from-trunk-sip-2231120 (I guess that's a custom context, else it's from-trunk)
    qualify=yes
    insercuce=very


    If the context from-trunk-sip-2231120 not exist, you couldn't call, and be called.

    If your operator needed a register string information, you must put it.

    Code:
    account:password@202.211.228.41
    To verify your conf, 2 commands:
    sip show peers
    sip show registry
    (in the case where you used the register string.)
     
  20. tsku

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    after add in the register string, sip show registry show status 'registered', however outgoing call still show circuit busy, with and without qualify and insecure in place.
     

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