register time out for sip

[shahi]

Here is details of my server:

goautodial-ce-2.1-final.iso | VERSION: 2.4-309a BUILD: 110430-1642 | Asterisk 1.4.39.1-vici | Single Server | No Digium/Sangoma Hardware | Installed squid and g729 codec | Intel(R) Pentium(R) CPU E5200 @ 2.50GHz | Dual Core

update server ip to 203.188.245.xx

have two lan

eth0 192.168.101.1 for agents(using proxy)
eth1 203.188.245.xx internet connection

carrier info

Registration String: register => 0267806237:xxxxxxxxxx@96.31.83.227:5060/0267806237

Account Entry:
[ECSsipOUT]
nat=yes
qualify=yes
trustrpid=yes
sendrpid=yes
username=0267806237
type=peer
secret=xxxxxxxxxx
host=96.31.83.227
fromuser=0267806237
context=trunkinbound
allow=g729
allow=gsm
allow=ulaw
insecure=very
canreinvite=no



Globals String: ECSsipOUT= SIP/ECSsipOUT

Dialplan Entry:

exten => _944X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _944X.,2,Dial(${ECSsipOUT}/${EXTEN:1},,tTor)
exten => _944X.,3,Hangup()


sip show registry
Host Username Refresh State Reg.Time
96.31.83.227:5060 0267806237 105 Registered Fri, 14 Sep 2012 05:12:16

sip show peers
Name/username Host Dyn Nat ACL Port Status
8020/8020 (Unspecified) D N 0 UNKNOWN
8019/8019 (Unspecified) D N 0 UNKNOWN
8018/8018 (Unspecified) D N 0 UNKNOWN
8017/8017 (Unspecified) D N 0 UNKNOWN
8016/8016 (Unspecified) D N 0 UNKNOWN
8015/8015 (Unspecified) D N 0 UNKNOWN
8014/8014 (Unspecified) D N 0 UNKNOWN
8013/8013 (Unspecified) D N 0 UNKNOWN
8012/8012 (Unspecified) D N 0 UNKNOWN
8011/8011 (Unspecified) D N 0 UNKNOWN
8010/8010 (Unspecified) D N 0 UNKNOWN
8009/8009 (Unspecified) D N 0 UNKNOWN
8008/8008 (Unspecified) D N 0 UNKNOWN
8007/8007 (Unspecified) D N 0 UNKNOWN
8006/8006 (Unspecified) D N 0 UNKNOWN
8005/8005 (Unspecified) D N 0 UNKNOWN
8004/8004 (Unspecified) D N 0 UNKNOWN
8003/8003 (Unspecified) D N 0 UNKNOWN
8002/8002 (Unspecified) D N 0 UNKNOWN
8001/8001 (Unspecified) D N 0 UNKNOWN
ECSsipOUT/0267806237 96.31.83.227 N 5060 OK (317 ms)
21 sip peers [Monitored: 1 online, 20 offline Unmonitored: 0 online, 0 offline]


while calling using x-lite 3 I got this error in cli

busy code 21
declined from 96.31.83.227
and x-lite showed forbidden status.


I have another question that my agents can get register to our server while they are at office and using the proxy.why i am unable to register to my own server using x-lite from my home. I got 408 registration failed.

here is the x-lite login setting

user name and authorization name 8001
pass:xxxxxxx
domain:203.188.245.xx
checked the register with domain
send outbound via domain

any idea what is wrong?

[williamconley]

1) Welcome to the Party!

2) Since you are new, we'll start here:

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) If this is a "fresh" install that has never worked, you should probably begin by installing Vicibox (from Vicibox.com) or GoAutoDial (from GoAutoDial.com) instead of VicidialNOW which is very old.

4) Use the Vicidial Manager's Manual to configure your system for use. It is available (free version has everything you need for a working system!) at EFLO.net. Begin at page one and do everything without skipping any pages or exercises. Let us know the page/paragraph and manual version you are using and we'll help you get going. (Try to use a manual version that matches your install if you can, but if you install 2.6 you'll need to use the 2.4 manual until the 2.6 manual comes out).

5) Next you'll need to post your Carrier information so we can attempt to resolve your issue if you still have it after your reinstall or using the manager's manual!

[shahi]

what is wrong i don't know. here is the cli

[Sep 15 09:40:16] Reliably Transmitting (NAT) to 96.31.83.227:5060:
INVITE sip:12053491280@96.31.83.227;cpd=on SIP/2.0
Via: SIP/2.0/UDP 203.188.245.38:5060;branch=z9hG4bK39b27e11;rport
From: "8019" <sip:4904327605@203.188.245.38>;tag=as05c8e16d
To: <sip:12053491280@96.31.83.227;cpd=on>
Contact: <sip:4904327605@203.188.245.38>
Call-ID: 4448aab86e31c4bb0266513a6ca46c85@203.188.245.38
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "8019" <sip:01750400444@203.188.245.38>;privacy=off;screen=no
Proxy-Authorization: Digest username="0267806237", realm="asterisk", algorithm=MD5, uri="sip:12053491280@96.31.83.227;cpd=on", nonce="1615b29b", response="7e06fc83337419527d95c88b919fef38"
Date: Sat, 15 Sep 2012 13:40:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 2841 2842 IN IP4 203.188.245.38
s=session
c=IN IP4 203.188.245.38
t=0 0
m=audio 17390 RTP/AVP 0 18 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 15 09:40:17]
<--- SIP read from 96.31.83.227:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 203.188.245.38:5060;branch=z9hG4bK39b27e11;received=203.188.245.38;rport=5060
From: "8019" <sip:4904327605@203.188.245.38>;tag=as05c8e16d
To: <sip:12053491280@96.31.83.227;cpd=on>;tag=as0b8b7850
Call-ID: 4448aab86e31c4bb0266513a6ca46c85@203.188.245.38
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------->
[Sep 15 09:40:17] --- (10 headers 0 lines) ---
[Sep 15 09:40:17] Transmitting (NAT) to 96.31.83.227:5060:
ACK sip:12053491280@96.31.83.227;cpd=on SIP/2.0
Via: SIP/2.0/UDP 203.188.245.38:5060;branch=z9hG4bK39b27e11;rport
From: "8019" <sip:4904327605@203.188.245.38>;tag=as05c8e16d
To: <sip:12053491280@96.31.83.227;cpd=on>;tag=as0b8b7850
Contact: <sip:4904327605@203.188.245.38>
Call-ID: 4448aab86e31c4bb0266513a6ca46c85@203.188.245.38
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "8019" <sip:01750400444@203.188.245.38>;privacy=off;screen=no
Content-Length: 0


---
[Sep 15 09:40:17] WARNING[3005]: chan_sip.c:13482 handle_response_invite: Received response: "Forbidden" from '"8019" <sip:4904327605@203.188.245.38>;tag=as05c8e16d'
[Sep 15 09:40:17] -- SIP/ECSsipOUT-00000034 is circuit-busy
[Sep 15 09:40:17] == Everyone is busy/congested at this time (1:0/1/0)
[Sep 15 09:40:17] -- Executing [12053491280@default:3] Hangup("SIP/8019-00000033", "") in new stack
[Sep 15 09:40:17] == Spawn extension (default, 12053491280, 3) exited non-zero on 'SIP/8019-00000033'
[Sep 15 09:40:17] -- Executing [h@default:1] DeadAGI("SIP/8019-00000033", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
[Sep 15 09:40:17] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 15 09:40:17] Scheduling destruction of SIP dialog '0608ec380f74f910YWY0NmE4YjI5ZDBjZGJmMjE0OWIyM2ZiZDIwNDA0YTA.' in 32000 ms (Method: INVITE)
[Sep 15 09:40:17]
<--- Reliably Transmitting (NAT) to 192.168.101.17:13290 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.101.17:13290;branch=z9hG4bK-d87543-187a2f6e00684f1c-1--d87543-;received=192.168.101.17;rport=13290
From: "8019"<sip:8019@203.188.245.38>;tag=ef49ce34
To: "12053491280"<sip:12053491280@203.188.245.38>;tag=as4bd0ef87
Call-ID: 0608ec380f74f910YWY0NmE4YjI5ZDBjZGJmMjE0OWIyM2ZiZDIwNDA0YTA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>

[williamconley]

<--- SIP read from 96.31.83.227:5060 --->
SIP/2.0 403 Forbidden

means your carrier refused to connect the call. contact the carrier and find out why