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.
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
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.
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 ??
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.
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.
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.
