Inbound calls - 603 Denied

Hello

I have been trying to configure inbound calls on my system, and it allways gives me 603 denied.

My DDI/DID wich is forwarding the calls to my domain at iptel.org.

Then the trunk is between my pbx and iptel.org

The trunk geisters all fine but when i call my DDI/DID i allways get 603 - Denied error.

What should i do ?

Here are the configurations of my system

[iptelinbound] type = friend host = iptel.org username = limitexxxxx secret = xxxxx context = incoming transport = udp nat = no qualify = yes canreinvite = no amaflags = default dtmfmode = rfc2833 insecure = port,invite disallow = all allow = ulaw trustrpid = no call-limit = 30 fromuser = limitexxxxx@iptel.org fromdomain = iptel.org

[general] disallow = all context = incoming ; ; ** realtime friends caching ** ; keep this on if you wish to use realtime peers ; with NAT and MWI (message waiting indicator) ; rtcachefriends = yes allowguest = no autocreatep = no canreinvite = yes dtmfmode = rfc2833 nat = no ignoreregexpire = no progressinband = no promiscredir = no srvlookup = no videosupport = yes trustrpid = no rtautoclear = no rtupdate = yes allow = ulaw tos_audio = ef externip = 192.168.1.104 match_auth_username = yes allowoverlap = yes allowtransfer = yes realm = ipbx bindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 tlsenable = no tlsbindaddr = 0.0.0.0 tlsdontverifyserver = no tlscipher = DEFAULT pedantic = no tos_sip = cs3 tos_video = af41 tos_text = af41 cos_sip = 3 cos_audio = 5 cos_video = 4 cost_text = 3 maxexpiry = 3600 minexpiry = 60 defaultexpiry = 120 mwiexpiry = 3600 qualifyfreq = 60 qualifygap = 100 qualifypeers = 5 notifymimetype = application/simple-message-summary buggymwi = no vmexten = ipbx mohinterpret = default mohsuggest = default parkinglot = plaza language = en relaxdtmf = no sendrpid = no prematuremedia = yes useragent = taridium ipbx sdpsession = taridium ipbx sdpowner = root usereqphone = no compactheaders = yes maxcallbitrate = 384 callevents = no authfailureevents = no alwaysauthreject = yes g726nonstandard = yes matchexterniplocally = yes dynamic_exclude_static = yes forwardloopdetected = yes shrinkcallerid = yes regextenonqualify = no t1min = 100 timert1 = 500 timerb = 32000 rtptimeout = 60 rtpholdtimeout = 300 rtpkeepalive = 0 session-timers = originate session-expires = 1800 session-minse = 90 session-refresher = uas sipdebug = no recordhistory = no dumphistory = no allowsubscribe = yes subscribecontext = subs notifyringing = yes notifyhold = no notifycid = no callcounters = yes t38pt_udptl = no faxdetect = no directmedia = yes directrtpsetup = no ignoresdpversion = no rtsavesysname = no allowexternaldomains = yes autodomain = no jbenable = no jbforce = no jbmaxsize = 200 jbresyncthreshold = 1000 jbimpl = fixed jbtargetextra = 40 jblog = no register = limitexxxxx:xxxxx:limitexxxxx@iptel.org

Some debug right after a failed call

> doing dnsmgr_lookup for 'iptel.org' == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [s@incoming:1] AGI("SIP/iptelinbound-00000011", "ipbx/incoming.ipbx") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/ipbx/incoming.ipbx > doing dnsmgr_lookup for 'iptel.org' > ipbx/incoming.ipbx: no incoming routing table entry for extension (s) -- <SIP/iptelinbound-00000011>AGI Script ipbx/incoming.ipbx completed, returning 4 == Spawn extension (incoming, s, 1) exited non-zero on 'SIP/iptelinbound-00000011' -- Executing [h@incoming:1] Hangup("SIP/iptelinbound-00000011", "") in new stack == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/iptelinbound-00000011' [May 14 17:55:32] NOTICE[2809]: chan_sip.c:26387 sip_poke_noanswer: Peer 'iptelinbound' is now UNREACHABLE! Last qualify: 71 ipbx*CLI>

that is all if any more information needed i would gladly provide

here is also some more complet debug output, this is when i try to use 664

[code]<— SIP read from UDP:192.168.1.101:50647 —>
REGISTER sip:192.168.1.104 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK07550298
From: sip:2002@192.168.1.104;tag=0015c65a535e00222440ea8f-3affa637
To: sip:2002@192.168.1.104
Call-ID: 0015c65a-535e0002-1a3f5536-18937793@192.168.1.101
Max-Forwards: 70
CSeq: 165 REGISTER
User-Agent: Cisco-CP7960G/8.0
Contact: sip:2002@192.168.1.101:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0015c65a535e”;+u.sip!model.ccm.cisco.com="7"
Content-Length: 0
Expires: 60

<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.101:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.101:5060 —>
SIP/2.0 401 Unauthorized
v: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK07550298;received=192.168.1.101
f: sip:2002@192.168.1.104;tag=0015c65a535e00222440ea8f-3affa637
t: sip:2002@192.168.1.104;tag=as6aa6c35e
i: 0015c65a-535e0002-1a3f5536-18937793@192.168.1.101
CSeq: 165 REGISTER
Server: taridium ipbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“ipbx”, nonce="00a8e1e4"
l: 0

<------------>
Scheduling destruction of SIP dialog ‘0015c65a-535e0002-1a3f5536-18937793@192.168.1.101’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.101:50647 —>
REGISTER sip:192.168.1.104 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK2f5b3bca
From: sip:2002@192.168.1.104;tag=0015c65a535e00222440ea8f-3affa637
To: sip:2002@192.168.1.104
Call-ID: 0015c65a-535e0002-1a3f5536-18937793@192.168.1.101
Max-Forwards: 70
CSeq: 166 REGISTER
User-Agent: Cisco-CP7960G/8.0
Contact: sip:2002@192.168.1.101:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0015c65a535e”;+u.sip!model.ccm.cisco.com="7"
Authorization: Digest username=“2002”,realm=“ipbx”,uri=“sip:192.168.1.104”,response=“e9edcdaf716ed81614266bbe288069ac”,nonce=“00a8e1e4”,algorithm=MD5
Content-Length: 0
Expires: 60

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.101:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.101:5060 —>
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK2f5b3bca;received=192.168.1.101
f: sip:2002@192.168.1.104;tag=0015c65a535e00222440ea8f-3affa637
t: sip:2002@192.168.1.104;tag=as6aa6c35e
i: 0015c65a-535e0002-1a3f5536-18937793@192.168.1.101
CSeq: 166 REGISTER
Server: taridium ipbx
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
Expires: 60
m: sip:2002@192.168.1.101:5060;transport=udp;expires=60
Date: Wed, 15 May 2013 08:58:26 GMT
l: 0

<------------>
Scheduling destruction of SIP dialog ‘378e80744c93efe4452f448d7dc3694a@192.168.1.104:5060’ in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.1.101:5060:
NOTIFY sip:2002@192.168.1.101:5060;transport=udp SIP/2.0
v: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK5ab704b1
Max-Forwards: 70
f: “asterisk” sip:asterisk@192.168.1.104;tag=as34201609
t: sip:2002@192.168.1.101:5060;transport=udp
m: sip:asterisk@192.168.1.104:5060
i: 378e80744c93efe4452f448d7dc3694a@192.168.1.104:5060
CSeq: 102 NOTIFY
User-Agent: taridium ipbx
o: message-summary
c: application/simple-message-summary
l: 89

Messages-Waiting: no
Message-Account: sip:ipbx@192.168.1.104
Voice-Message: 0/0 (0/0)


Scheduling destruction of SIP dialog ‘0015c65a-535e0002-1a3f5536-18937793@192.168.1.101’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.101:50688 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK5ab704b1
From: “asterisk” sip:asterisk@192.168.1.104;tag=as34201609
To: sip:2002@192.168.1.101:5060;transport=udp
Call-ID: 378e80744c93efe4452f448d7dc3694a@192.168.1.104:5060
CSeq: 102 NOTIFY
Content-Length: 0
[/code]

The problem is not in Asterisk but in the AGI script that you are running. Start with debugging there.

So could you guide me how to do that ? i am using Freepbx, I installed the distro on their website, clean install just extensions and 2 IP-Phones configured.

Taridium’s contact web page is taridium.com/corp-en/?page=contact

We know nothing about how their AGI script works.

One thing to note is that fromuser should not contain @, but that would not cause the problems you are seeing.

Sorry it was my mistake there, i have two machines one with taridium and another with freepbx, im using the one with freepbx. :confused: i edited my last comment but it was too late.

once again sorry for my mistake

Best Regards

Ricardo Costa

freepbx.org/forums/

Thank you.

Sory to make you waste your time with this problem.

once again thanks

Best Regards

Ricardo Costa