CAN NOT RECEIVE CALLS FROM SIP TRUNK

[osmose]

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.

 

[Bob]

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

 

[fmvillares]

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

 

[osmose]

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.

 

[fmvillares]

Re: Re:CAN NOT RECEIVE CALLS FROM SIP TRUNK

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

 

[siptellnet]

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 ??

 

[dicko]

Re: Re:CAN NOT RECEIVE CALLS FROM SIP TRUNK

I suggest you might possibly be using illegimate codecs.

 

[siptellnet]

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.

 

[dicko]

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.

 

[johnatan]

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.