CAN NOT RECEIVE CALLS FROM SIP TRUNK

Discussion in 'General' started by osmose, Jan 7, 2011.

  1. osmose

    Joined:
    May 29, 2010
    Messages:
    13
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    0
    Hello everybody

    I can not receive calls from my SIP provider.

    I have tried with different extensions, with or without ring groups, deleting and recreating extension, trunk and inbound route etc still no solution.

    My problem is that I can not understand what is happening and where should I search for the error.

    I am appending a few details and a log that if anybody wants to go through maybe helpfull.

    I am open to any suggestions.

    My Elastix is registered with the provider with a sip trunk.

    ----
    pbx*CLI> sip show registry
    Host dnsmgr Username Refresh State Reg.Time
    voip.viva.gr:5060 N 302118002553 105 Registered Sat, 08 Jan 2011 00:38:03
    ----

    All telephones (UA) are registered

    ----
    pbx*CLI> sip show peers
    Name/username Host Dyn Nat ACL Port Status
    500/500 10.1.10.97 D N A 5063 OK (71 ms)
    5002/5002 10.1.10.96 D N A 6000 OK (77 ms)
    501/501 10.1.10.85 D N A 5062 OK (53 ms)
    502/502 10.1.10.96 D N A 5064 OK (166 ms)
    505/505 10.1.10.52 D N A 5062 OK (67 ms)
    506/506 10.1.10.52 D N A 5063 OK (61 ms)
    507/507 10.1.10.97 D N A 5065 OK (72 ms)
    508/508 10.1.10.96 D N A 5062 OK (76 ms)
    600/600 10.1.10.97 D N A 5062 OK (75 ms)
    601/601 10.1.10.85 D N A 5063 OK (65 ms)
    602/602 10.1.10.96 D N A 5063 OK (151 ms)
    630/630 10.1.10.97 D N A 5066 OK (78 ms)
    632/632 10.1.10.52 D N A 5064 OK (50 ms)
    633/633 10.1.10.96 D N A 5065 OK (76 ms)
    650/650 10.1.10.97 D N A 5064 OK (67 ms)
    651/651 10.1.10.85 D N A 5064 OK (65 ms)
    viva-sip-trunk/3021180025 83.235.24.86 N 5060 OK (24 ms)

    My sip settings are as follows:

    ----

    Global Settings:
    ----------------
    UDP SIP Port: 5060
    UDP Bindaddress: 0.0.0.0
    TCP SIP Port: Disabled
    TLS SIP Port: Disabled
    Videosupport: No
    Textsupport: No
    Ignore SDP sess. ver.: No
    AutoCreate Peer: No
    Match Auth Username: No
    Allow unknown access: Yes
    Allow subscriptions: Yes
    Allow overlap dialing: Yes
    Allow promsic. redir: No
    Enable call counters: No
    SIP domain support: No
    Realm. auth: No
    Our auth realm asterisk
    Call to non-local dom.: Yes
    URI user is phone no: No
    Always auth rejects: Yes
    Direct RTP setup: No
    User Agent: Asterisk PBX 1.6.2.10
    SDP Session Name: Asterisk PBX 1.6.2.10
    SDP Owner Name: root
    Reg. context: (not set)
    Regexten on Qualify: No
    Caller ID: Unknown
    From: Domain:
    Record SIP history: Off
    Call Events: Off
    Auth. Failure Events: Off
    T.38 support: No
    T.38 EC mode: Unknown
    T.38 MaxDtgrm: -1
    SIP realtime: Disabled
    Qualify Freq : 60000 ms

    Network QoS Settings:
    ---------------------------
    IP ToS SIP: CS3
    IP ToS RTP audio: EF
    IP ToS RTP video: AF41
    IP ToS RTP text: CS0
    802.1p CoS SIP: 4
    802.1p CoS RTP audio: 5
    802.1p CoS RTP video: 6
    802.1p CoS RTP text: 5
    Jitterbuffer enabled: No
    Jitterbuffer forced: No
    Jitterbuffer max size: -1
    Jitterbuffer resync: -1
    Jitterbuffer impl:
    Jitterbuffer log: No

    Network Settings:
    ---------------------------
    SIP address remapping: Enabled using externhost
    Externhost: xxxxxxxxxx.homelinux.net
    Externip: 194.618.249.213:5060
    Externrefresh: 120
    Internal IP: 127.0.0.1:5060
    Localnet: 10.1.10.0/255.255.255.0
    STUN server: 0.0.0.0:0

    Global Signalling Settings:
    ---------------------------
    Codecs: 0xe (gsm|ulaw|alaw)
    Codec Order: ulaw:20,gsm:20,alaw:20
    Relax DTMF: No
    RFC2833 Compensation: No
    Compact SIP headers: No
    RTP Keepalive: 0 (Disabled)
    RTP Timeout: 30
    RTP Hold Timeout: 300
    MWI NOTIFY mime type: application/simple-message-summary
    DNS SRV lookup: No
    Pedantic SIP support: No
    Reg. min duration 60 secs
    Reg. max duration: 3600 secs
    Reg. default duration: 120 secs
    Outbound reg. timeout: 20 secs
    Outbound reg. attempts: 0
    Notify ringing state: Yes
    Include CID: No
    Notify hold state: Yes
    SIP Transfer mode: open
    Max Call Bitrate: 384 kbps
    Auto-Framing: No
    Outb. proxy: <not set>
    Session Timers: Accept
    Session Refresher: uas
    Session Expires: 1800 secs
    Session Min-SE: 90 secs
    Timer T1: 500
    Timer T1 minimum: 100
    Timer B: 32000
    No premature media: Yes

    Default Settings:
    -----------------
    Allowed transports: UDP
    Outbound transport: UDP
    Context: from-sip-external
    Nat: Always
    DTMF: rfc2833
    Qualify: 0
    Use ClientCode: No
    Progress inband: Never
    Language:
    MOH Interpret: default
    MOH Suggest:
    Voice Mail Extension: *97

    ----

    Though when I am making an incoming call the asterisk log shows:

    -----
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    -- Executing [302118002553@from-trunk:1] Set("SIP/VIVA-00000000", "__FROM_DID=302118002553") in new stack
    -- Executing [302118002553@from-trunk:2] Gosub("SIP/VIVA-00000000", "app-blacklist-check,s,1") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/VIVA-00000000", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/VIVA-00000000", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/VIVA-00000000", "") in new stack
    -- Executing [302118002553@from-trunk:3] ExecIf("SIP/VIVA-00000000", "0 ?Set(CALLERID(name)=6973079088)") in new stack
    -- Executing [302118002553@from-trunk:4] Set("SIP/VIVA-00000000", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [302118002553@from-trunk:5] Set("SIP/VIVA-00000000", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [302118002553@from-trunk:6] Goto("SIP/VIVA-00000000", "from-did-direct,500,1") in new stack
    -- Goto (from-did-direct,500,1)
    -- Executing [500@from-did-direct:1] Macro("SIP/VIVA-00000000", "exten-vm,500,500") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/VIVA-00000000", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/VIVA-00000000", "AMPUSER=6973079088") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/VIVA-00000000", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/VIVA-00000000", "1?Set(REALCALLERIDNUM=6973079088)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/VIVA-00000000", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/VIVA-00000000", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/VIVA-00000000", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/VIVA-00000000", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/VIVA-00000000", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/VIVA-00000000", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/VIVA-00000000", "Using CallerID "306973079088" <6973079088>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/VIVA-00000000", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/VIVA-00000000", "VMBOX=500") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/VIVA-00000000", "__EXTTOCALL=500") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/VIVA-00000000", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/VIVA-00000000", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/VIVA-00000000", "RT=30") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/VIVA-00000000", "record-enable,500,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/VIVA-00000000", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/VIVA-00000000", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/VIVA-00000000", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/VIVA-00000000", "1?IN") in new stack
    -- Goto (macro-record-enable,s,20)
    -- Executing [s@macro-record-enable:20] ExecIf("SIP/VIVA-00000000", "1?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/VIVA-00000000", "dial-one,30,tr,500") in new stack
    -- Executing [s@macro-dial-one:1] Set("SIP/VIVA-00000000", "DEXTEN=500") in new stack
    -- Executing [s@macro-dial-one:2] Set("SIP/VIVA-00000000", "DIALSTATUS_CW=") in new stack
    -- Executing [s@macro-dial-one:3] GosubIf("SIP/VIVA-00000000", "0?screen,1") in new stack
    -- Executing [s@macro-dial-one:4] GosubIf("SIP/VIVA-00000000", "0?cf,1") in new stack
    -- Executing [s@macro-dial-one:5] GotoIf("SIP/VIVA-00000000", "1?skip1") in new stack
    -- Goto (macro-dial-one,s,8)
    -- Executing [s@macro-dial-one:8] GotoIf("SIP/VIVA-00000000", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:9] GotoIf("SIP/VIVA-00000000", "0?continue") in new stack
    -- Executing [s@macro-dial-one:10] Set("SIP/VIVA-00000000", "EXTHASCW=") in new stack
    -- Executing [s@macro-dial-one:11] GotoIf("SIP/VIVA-00000000", "1?next1:cwinusebusy") in new stack
    -- Goto (macro-dial-one,s,12)
    -- Executing [s@macro-dial-one:12] GotoIf("SIP/VIVA-00000000", "0?docfu:skip3") in new stack
    -- Goto (macro-dial-one,s,16)
    -- Executing [s@macro-dial-one:16] GotoIf("SIP/VIVA-00000000", "1?next2:continue") in new stack
    -- Goto (macro-dial-one,s,17)
    -- Executing [s@macro-dial-one:17] GotoIf("SIP/VIVA-00000000", "1?continue") in new stack
    -- Goto (macro-dial-one,s,25)
    -- Executing [s@macro-dial-one:25] GotoIf("SIP/VIVA-00000000", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:26] GosubIf("SIP/VIVA-00000000", "1?dstring,1:dlocal,1") in new stack
    -- Executing [dstring@macro-dial-one:1] Set("SIP/VIVA-00000000", "DSTRING=") in new stack
    -- Executing [dstring@macro-dial-one:2] Set("SIP/VIVA-00000000", "DEVICES=500") in new stack
    -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/VIVA-00000000", "0?Return()") in new stack
    -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/VIVA-00000000", "0?Set(DEVICES=00)") in new stack
    -- Executing [dstring@macro-dial-one:5] Set("SIP/VIVA-00000000", "LOOPCNT=1") in new stack
    -- Executing [dstring@macro-dial-one:6] Set("SIP/VIVA-00000000", "ITER=1") in new stack
    -- Executing [dstring@macro-dial-one:7] Set("SIP/VIVA-00000000", "THISDIAL=SIP/500") in new stack
    -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/VIVA-00000000", "1?zap2dahdi,1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/VIVA-00000000", "0?Return()") in new stack
    -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/VIVA-00000000", "NEWDIAL=") in new stack
    -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/VIVA-00000000", "LOOPCNT2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/VIVA-00000000", "ITER2=1") in new stack
    -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/VIVA-00000000", "THISPART2=SIP/500") in new stack
    -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/VIVA-00000000", "0?Set(THISPART2=DAHDI/500)") in new stack
    -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/VIVA-00000000", "NEWDIAL=SIP/500&") in new stack
    -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/VIVA-00000000", "ITER2=2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/VIVA-00000000", "0?begin2") in new stack
    -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/VIVA-00000000", "THISDIAL=SIP/500") in new stack
    -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/VIVA-00000000", "") in new stack
    -- Executing [dstring@macro-dial-one:9] Set("SIP/VIVA-00000000", "DSTRING=SIP/500&") in new stack
    -- Executing [dstring@macro-dial-one:10] Set("SIP/VIVA-00000000", "ITER=2") in new stack
    -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/VIVA-00000000", "0?begin") in new stack
    -- Executing [dstring@macro-dial-one:12] Set("SIP/VIVA-00000000", "DSTRING=SIP/500") in new stack
    -- Executing [dstring@macro-dial-one:13] Return("SIP/VIVA-00000000", "") in new stack
    -- Executing [s@macro-dial-one:27] GotoIf("SIP/VIVA-00000000", "0?nodial") in new stack
    -- Executing [s@macro-dial-one:28] GotoIf("SIP/VIVA-00000000", "1?skiptrace") in new stack
    -- Goto (macro-dial-one,s,30)
    -- Executing [s@macro-dial-one:30] Set("SIP/VIVA-00000000", "D_OPTIONS=tr") in new stack
    -- Executing [s@macro-dial-one:31] ExecIf("SIP/VIVA-00000000", "0?SIPAddHeader(Alert-Info: )") in new stack
    -- Executing [s@macro-dial-one:32] ExecIf("SIP/VIVA-00000000", "0?SIPAddHeader()") in new stack
    -- Executing [s@macro-dial-one:33] ExecIf("SIP/VIVA-00000000", "0?SetMusicOnHold()") in new stack
    -- Executing [s@macro-dial-one:34] GosubIf("SIP/VIVA-00000000", "0?qwait,1") in new stack
    -- Executing [s@macro-dial-one:35] Set("SIP/VIVA-00000000", "__CWIGNORE=") in new stack
    -- Executing [s@macro-dial-one:36] Set("SIP/VIVA-00000000", "__KEEPCID=TRUE") in new stack
    -- Executing [s@macro-dial-one:37] Dial("SIP/VIVA-00000000", "SIP/500,30,tr") in new stack
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    -- Couldn't call 500
    == Everyone is busy/congested at this time (0:0/0/0)
    -- Executing [s@macro-dial-one:38] ExecIf("SIP/VIVA-00000000", "0?Set(DIALSTATUS=)") in new stack
    -- Executing [s@macro-dial-one:39] GosubIf("SIP/VIVA-00000000", "0?s-CHANUNAVAIL,1") in new stack
    -- Executing [s@macro-dial-one:40] MacroExit("SIP/VIVA-00000000", "") in new stack
    -- Executing [s@macro-exten-vm:10] GotoIf("SIP/VIVA-00000000", "0?exit") in new stack
    -- Executing [s@macro-exten-vm:11] Set("SIP/VIVA-00000000", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:12] GosubIf("SIP/VIVA-00000000", "0?docfu,1") in new stack
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/VIVA-00000000", "0?docfb,1") in new stack
    -- Executing [s@macro-exten-vm:14] Set("SIP/VIVA-00000000", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:15] NoOp("SIP/VIVA-00000000", "Voicemail is '500'") in new stack
    -- Executing [s@macro-exten-vm:16] GotoIf("SIP/VIVA-00000000", "0?s-CHANUNAVAIL,1") in new stack
    -- Executing [s@macro-exten-vm:17] NoOp("SIP/VIVA-00000000", "Sending to Voicemail box 500") in new stack
    -- Executing [s@macro-exten-vm:18] Macro("SIP/VIVA-00000000", "vm,500,CHANUNAVAIL,") in new stack
    -- Executing [s@macro-vm:1] Macro("SIP/VIVA-00000000", "user-callerid,SKIPTTL") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/VIVA-00000000", "AMPUSER=6973079088") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/VIVA-00000000", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/VIVA-00000000", "0?Set(REALCALLERIDNUM=6973079088)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/VIVA-00000000", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/VIVA-00000000", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/VIVA-00000000", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/VIVA-00000000", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/VIVA-00000000", "Using CallerID "306973079088" <6973079088>") in new stack
    -- Executing [s@macro-vm:2] Set("SIP/VIVA-00000000", "VMGAIN=""") in new stack
    -- Executing [s@macro-vm:3] GotoIf("SIP/VIVA-00000000", "1?vmx,1") in new stack
    -- Goto (macro-vm,vmx,1)
    -- Executing [vmx@macro-vm:1] Set("SIP/VIVA-00000000", "MEXTEN=500") in new stack
    -- Executing [vmx@macro-vm:2] Set("SIP/VIVA-00000000", "MMODE=CHANUNAVAIL") in new stack
    -- Executing [vmx@macro-vm:3] Set("SIP/VIVA-00000000", "RETVM=") in new stack
    -- Executing [vmx@macro-vm:4] Set("SIP/VIVA-00000000", "MODE=unavail") in new stack
    -- Executing [vmx@macro-vm:5] GotoIf("SIP/VIVA-00000000", "1?chknomsg") in new stack
    -- Goto (macro-vm,vmx,7)
    -- Executing [vmx@macro-vm:7] GotoIf("SIP/VIVA-00000000", "0?s-CHANUNAVAIL,1") in new stack
    -- Executing [vmx@macro-vm:8] GotoIf("SIP/VIVA-00000000", "1?notdirect") in new stack
    -- Goto (macro-vm,vmx,10)
    -- Executing [vmx@macro-vm:10] NoOp("SIP/VIVA-00000000", "Checking if ext 500 is enabled: ") in new stack
    -- Executing [vmx@macro-vm:11] GotoIf("SIP/VIVA-00000000", "1?s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-vm,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-vm:1] Macro("SIP/VIVA-00000000", "get-vmcontext,500") in new stack
    -- Executing [s@macro-get-vmcontext:1] Set("SIP/VIVA-00000000", "VMCONTEXT=default") in new stack
    -- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/VIVA-00000000", "0?200:300") in new stack
    -- Goto (macro-get-vmcontext,s,300)
    -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/VIVA-00000000", "") in new stack
    -- Executing [s-CHANUNAVAIL@macro-vm:2] VoiceMail("SIP/VIVA-00000000", "500@default,u") in new stack
    == Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/VIVA-00000000' in macro 'vm'
    == Spawn extension (macro-exten-vm, s, 18) exited non-zero on 'SIP/VIVA-00000000' in macro 'exten-vm'
    == Spawn extension (from-did-direct, 500, 1) exited non-zero on 'SIP/VIVA-00000000'
    -- Executing [h@from-did-direct:1] Macro("SIP/VIVA-00000000", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/VIVA-00000000", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/VIVA-00000000", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/VIVA-00000000", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/VIVA-00000000", "") in new stack
    == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/VIVA-00000000' in macro 'hangupcall'
    == Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/VIVA-00000000'

    --------------------------

    PRODUCTS VERSION

    Name Package Name Version Release
    Kernel
    Linux(x86_64) 2.6.18 194.3.1.el5
    Name Package Name Version Release
    Elastix
    elastix 2.0.0 58
    elastix-asterisk-sounds 1.2.3 1
    elastix-firstboot 2.0.0 10
    elastix-email_admin 2.0.0 16
    elastix-system 2.0.0 26
    elastix-vtigercrm 5.1.0 8
    elastix-agenda 2.0.0 14
    elastix-fax 2.0.0 12
    elastix-reports 2.0.0 14
    elastix-a2billing 1.3.0 4
    elastix-addons 2.0.0 14
    elastix-pbx 2.0.0 22
    Name Package Name Version Release
    RounCubeMail
    RoundCubeMail 0.3.1 4
    Name Package Name Version Release
    Mail
    postfix 2.3.3 2.1.el5_2
    cyrus-imapd 2.3.7 7.el5_4.3
    Name Package Name Version Release
    IM
    openfire 3.5.1 3
    Name Package Name Version Release
    FreePBX
    freePBX 2.7.0 5beta
    Name Package Name Version Release
    Asterisk
    asterisk 1.6.2.10 1
    asterisk-perl 0.10 2
    asterisk-addons 1.6.2.1 0
    Name Package Name Version Release
    FAX
    hylafax 4.3.9 0rhel5
    iaxmodem 1.2.0 1.1
    Name Package Name Version Release
    DRIVERS
    dahdi 2.3.0.1 3
    rhino 0.99.3 2.beta2
    wanpipe-util 3.5.14 0

    Thanks a lot for your help.
     
  2. Bob

    Bob

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    Thank you for providing a reasonable amount of detail, it makes it easier to look at the issue.

    One more thing is important, especially when dealing with SIP trunks is to provide the Trunk configuration, which we need to provide some reasonable answers.

    Taking a guess however, you have either the context set wrong in the trunk settings or you have a codec negotiation issue (e.g you have set G729 and you haven't got the codec installed - set to ULAW to keep things simple for the moment)...however I would concentrate on the context first, unless the codec issue rings a bell.

    Regards

    Bob
     
  3. fmvillares

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    as bob said
    disallow=all
    allow=ulaw,alaw ---- only those 2 for the test
    context=from-trunk
     
  4. osmose

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    The trunk configurations is as follows:

    context=from-trunk
    fromuser=302118001111
    username=302118001111
    secret=xxxxxxxxxxxx
    host=telephony.viva.gr
    srvlookup=yes
    insecure=port,invite
    canreinvite=no
    dtmfmode=rfc2833
    t38pt_udptl=yes
    nat=yes
    qualify=yes
    type=peer
    disallow=all
    allow=alaw&ulaw

    I have checked those 2 codecs alaw and ulaw at the ip phones too.
     
  5. fmvillares

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    Re: Re:CAN NOT RECEIVE CALLS FROM SIP TRUNK

    type=friend... peer is for sending calls only
     
  6. siptellnet

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    Re: Re:CAN NOT RECEIVE CALLS FROM SIP TRUNK

    I did a clean installation of Elastix 2.0 64 Bits.
    I have a similar problem, outbound calls works perfect, but not incomming calls from sip-trunk. The call doesn't ring
    All ports are open, I had working a 1.6 version
    I receive this log, wht this mean??
    Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    == Using UDPTL TOS bits 184
    == Using UDPTL CoS mark 5
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    == Using UDPTL TOS bits 184
    == Using UDPTL CoS mark 5
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    == Using UDPTL TOS bits 184
    == Using UDPTL CoS mark 5
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    == Using UDPTL TOS bits 184
    == Using UDPTL CoS mark 5
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    == Using UDPTL TOS bits 184
    == Using UDPTL CoS mark 5


    Any idea ??
     
  7. dicko

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    Re: Re:CAN NOT RECEIVE CALLS FROM SIP TRUNK

    I suggest you might possibly be using illegimate codecs.
     
  8. siptellnet

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    Re: Re:CAN NOT RECEIVE CALLS FROM SIP TRUNK

    WRONG!!!!!Do not assume stupid things.
    Just installed the version 2 as it comes and did an yum -y update
    Please if you are not willing to provide fresh ideas, do not disturb the forum.
     
  9. dicko

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    Re: Re:CAN NOT RECEIVE CALLS FROM SIP TRUNK

    I assumed nothing, I merely suggested, you will just see those entries in your log if 'some' codec's are being renegotiated unsuccessfully.

    I guess I touched a tender spot. Fear not, I will no longer disturb your singular presences here in this forum.
     
  10. johnatan

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    Hi there,

    In the 2.x release you need to create 2 trunk.
    One for peer another for user.
    It is a stupid thing but never worked for me in one trunk.

    Try it with 2 trunk.
     

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