vicidial.org

can somebody please help me...

im using vicibox on an Intel quad core 4GB RAM 500GB Disk

my SIP/Account Entry on Carrier Settings
[AccelaSansay1]
type=peer
nat=yes
host=69.94.226.174
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
dtmfmode=inband


my extensions/Dial Plan Entry
Global String:TRUNKY = SIP/AccelaSansay1
exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91999NXXXXXX,2,Dial(${TRUNKY}/${EXTEN:2},,tTor
exten => _91999NXXXXXX,3,Hangup

Outbound is working fine, only problem is INBOUND

Nothing happens on Asterisk during inbound call
Nothing displayed entering "sip show subscriptions"

I managed to get the log from our VOIP Provider..here's the log
////////////////////////
Session 194-1271227014@69.94.226.174, Release Cause (code) = 0402
T Stack Cause = 404, Not Found
Start = Wed Apr 14 06:36:54 2010
Answer = NA
Release = Wed Apr 14 06:36:55 2010
Duration = 0
Post Dial Delay = 1
Ring Time = 0

Term TID = 000389, Proc_SIP
ANI = 2142702102, From = 69.94.226.174
DNIS = 8778880524, To = 112.202.42.233
Call Leg ID = 194-1-1271227014@69.94.226.174
Media IP:UDP = 0.0.0.0:0
Switched IP:UDP = 69.94.226.174:33534
Codec List = NA ////////////////////


I followed the managers manual from creating the Inbound group and pointing a DID into it but nothing is working..

i understand the the sip settings on vicibox is on sip-vicidial.conf and the extensions are basically on extensions-vicidial.conf

i might need to add something on these files but i cant get hold of the missing part...please help

admin.php version and build?

Why have you not set a context of "trunkinbound" for your sip trunk?

thanks for ur reply matt..
heres my admin.php build


VERSION: 2.0.5-173
BUILD: 90320-0424

after reading some posts here with the same problem...ive tried adding a context=trunkinbound

trunkinbound looks like this>>

[trunkinbound]
exten => 2345,1,Answer
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup

;DID routing
exten => _X.,1,AGI(agi-DID_route.agi)


please help me matt...

Asterisk CLI output?

Hi Matt,

Heres my comprehensive overall server settings.
Can u please take a look and see if I made a mistake?



VICIBOX VERSION: 2.0.5-173
BUILD: 90320-0424
===================================================
Server specs: Intel QuadCore2/4GB RAM/500GB DISK/

Phone: X-lite from Counterpath
===================================================
Carrier name:AccelaSansay1
Carrier Record
Account Entry:
[AccelaSansay1]
type=friend
nat=yes
host=69.94.226.174
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
dtmfmode=inband
context=trunkinbound


Protocol:SIP
Globals String:TRUNKY = SIP/AccelaSansay1

=========================================

DID Extension:8778880524

=========================================

PHONES>>>

Phone Extension:201
Extension Context:trunkinbound
Outbound Call Group:SIP/AccelaSansay1



===========================================
EXTENSIONS.CONF>>>


[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNKY=SIP/AccelaSansay1
TRUNK=SIP/AccelaSansay1
TRUNKX=SIP/AccelaSansay1
;TRUNK=Zap/g1 ; Trunk interface
;TRUNKX=Zap/g2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk
SIPtrunk=SIP/202:1234@192.168.1.123
TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind monitoring
TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing

#include extensions-vicidial.conf

[trunkinbound]
; agent dial-in:
exten => 2345,1,Answer ; Answer the line
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup

exten => 201,1,Dial(SIP/202,,120)
exten => 201,2,Hangup

; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

===========================================

EXTENSIONS-VICIDIAL.CONF>>>


TRUNKY = SIP/AccelaSansay1

[vicidial-auto]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... ----------)

; Local Server: 192.168.1.123
exten => _192*168*001*123*.,1,Goto(default,${EXTEN:16},1)
; VICIDIAL Carrier: accella - AccelaSansay1
exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91999NXXXXXX,2,Dial(${TRUNKY}/${EXTEN:2},,tTor
exten => _91999NXXXXXX,3,Hangup


exten => 201,1,Dial(SIP/201)
exten => 201,2,Voicemail,u201
exten => 202,1,Dial(SIP/202)
exten => 202,2,Voicemail,u202



============================================
SIP.CONF>>>

[general]
context=trunkinbound ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
;realm=mydomain.tld ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld ; Set default domain for this host
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4 ; Add IP address as local domain
;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
;autodomain=yes ; Turn this on to have Asterisk add local host
;pedantic=yes ; Enable slow, pedantic checking for Pingtel
;tos=184 ; Set IP QoS to either a keyword or numeric val
tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
maxexpiry=3600 ; Max length of incoming registration we allow
defaultexpiry=120 ; Default length of incoming/outgoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
;videosupport=yes ; Turn on support for SIP video
;recordhistory=yes ; Record SIP history by default
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
;allow=alaw ;
musicclass=default ; Sets the default music on hold class for all SIP calls
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change the user agent string
promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
;dtmfmode = inband
;compactheaders = yes ; send compact sip headers.
;sipdebug = yes ; Turn on SIP debugging by default, from
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
;notifyringing = yes ; Notify subscriptions on RINGING state
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
;regcontext=sipregistrations
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
callevents=no ; generate manager events when sip ua performs events (e.g. hold)
;externip = 192.168.1.1 ; Address that we're going to put in outbound SIP messages
externip = 112.202.42.233
localnet = 192.168.1.123/255.255.255.0
;externhost=foo.dyndns.net ; Alternatively you can specify an
;externrefresh=10 ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=yes ; Global NAT settings (Affects all peers and users)
;nat=no
canreinvite=no
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes ; Enabling this setting has two functions:
; domain=myasterisk.dom
; domain=customer.com,customer-context
; autodomain=yes
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to

#include sip-vicidial.conf



==========================================================================
SIP-VICIDIAL.CONF>>>



; VICIDIAL Carrier: accella - AccelaSansay1
[AccelaSansay1]
type=friend
nat=yes
host=69.94.226.174
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
dtmfmode=inband
context=trunkinbound


[201]
username=201
secret=1234
mailbox=201
context=default
type=friend
host=dynamic

[202]
username=202
secret=1234
mailbox=202
context=default
type=friend
host=dynamic
=============================================================

Asterisk CLI output when a call comes in?

nothing happens on the asterisk cli matt during inbound call......i already enabled "sip debug"....

when i tried calling the number with our ACN videobox, i hear a message saying " Sorry, the number you dialed is not available..."

btw matt: here's our VOIP log during the inbound call....(i also tried asking them to give me a test call):
========================
Session 194-1271227014@69.94.226.174, Release Cause (code) = 0402
T Stack Cause = 404, Not Found
Start = Wed Apr 14 06:36:54 2010
Answer = NA
Release = Wed Apr 14 06:36:55 2010
Duration = 0
Post Dial Delay = 1
Ring Time = 0

Term TID = 000389, Proc_SIP
ANI = 2142702102, From = 69.94.226.174
DNIS = 8778880524, To = 112.202.42.233
Call Leg ID = 194-1-1271227014@69.94.226.174
Media IP:UDP = 0.0.0.0:0
Switched IP:UDP = 69.94.226.174:33534
Codec List = NA ////////////////////

Are you registering to your SIP provider?

Have you contacted them about this(since this is not a ViciDial issue)

1) if you do not register, they do not know to send the call to you

2) if they do not allow registry (and often even if they do!) there is often a location on their site to send the calls for this DID to a specific IP address

3) is your firewall open to allow the inbound call?

4) i have also had experience with a couple providers that after a DID was dead for XX minutes, it would never come back to life until we contacted the provider to reactivate it. (Server's power supply died, it took an hour to get a new one installed and get the system up ... when we did there were no inbound calls)

If there is NO activity in SIP debug when you dial the number, the call is not getting to your server. Vicidial and asterisk cannot fix this.

sorry for the late reply...im in cebu island in the philippines..
i just called my provider this morning regarding the registration string and they told me that there's no need for one..they only allowed our IP address as a form of authentication on their end...


my current changes on the settings is the context, i have audio problem using the trunkinbound context,but when i switched back to "default" context, outboud audio is fixed...wow! im really confused now..

im still having excellent outbound service....but no inbound service...i tried calling my vicidial server using our ACN videobox and the answer is:"Sorry, the number you dialled is not in service..."

another message when calling the server is "Were sorry, your call cannot be completed at this time, please hangup and place your call later."


Heres my asterisk log...theres nothing:

root@linux:~# asterisk -r
Asterisk 1.2.30.4, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.30.4 currently running on linux (pid = 6209)
-- Remote UNIX connection
Verbosity is at least 21
== Manager 'sendcron' logged off from 127.0.0.1
linux*CLI>
<-- SIP read from 192.168.1.102:7612:



--- (0 headers 1 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
linux*CLI>
<-- SIP read from 192.168.1.102:7612:



--- (0 headers 1 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
linux*CLI>
<-- SIP read from 192.168.1.102:7612:



--- (0 headers 1 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
linux*CLI>
<-- SIP read from 192.168.1.102:7612:



--- (0 headers 1 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
linux*CLI>
<-- SIP read from 192.168.1.102:7612:



--- (0 headers 1 lines) ---

williamconley wrote:3) is your firewall open to allow the inbound call?

if a call arrives on port 5060 and your firewall does not have 5060 pointed to your dialer ... your dialer will not get the call. (for starters)

hi william,

heres my router port settings:


single port forwarding: port 5060 both UDP/TCP>>forwarded to 192.168.1.123(Vicidial Server)

port range forwarding: ports 5000-20000 UDP >>forwarded to 192.168.1.123(Vicidial Server)

if you have 5060 forwarded to your server, and your server is not blocking 5060 via iptables or some other local firewall, and your SIP is set up standard and is listening on port 5060, and there is NO traffic in SIP when you call your DID ... the only logical conclusion is that the DID is not being forwarded to your server from your provider.

are you SURE you are not getting ANY traffic on your sip debug when you call the DID?

a message stating "no service" is actually NORMAL (it's the default message played by vicidial for an inbound DID unless you change it to go to an ingroup or phone or extension).

BUT: when the message stating no service is played to the caller, you SHOULD get sip traffic AND the asterisk CLI should even show the ss-noservice message being played to the caller.

hi william,

im sure there is no activity on cli and theres no firewall blocking forwarded packets from router.

i just called our voip provider and told them the possible problem on their end--calls not being forwarded...

they confirmed on the possible error on their end, theyre checking it out right now...

hi guys,

Our VOIP provider confirmed that after thorough investigation, everything is alright on their end..

William/Matt, I would like to ask what's the most essential line for the context in extensions.conf for the inbound call to work?

I think my mistake is in these area.

ive been making a lot of changes on the settings lately and ive this log on cli during an inbound call in progress >>

Executing AGI("SIP/69.94.226.174-08205f10", "agi://127.0.0.1:4577//call_log") in new stack
-- AGI Script agi://127.0.0.1:4577//call_log completed, returning 0
-- Executing Dial("SIP/69.94.226.174-08205f10", "sip/gs102|55|o") in new stack
Apr 20 11:32:11 NOTICE[25729]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/69.94.226.174-08205f10", "") in new stack
== Spawn extension (default, 8778880524, 3) exited non-zero on 'SIP/69.94.226.174-08205f10'
-- Executing DeadAGI("SIP/69.94.226.174-08205f10", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
> Channel SIP/201-081ff4d8 was answered.
-- Executing MeetMe("SIP/201-081ff4d8", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
== Manager 'sendcron' logged off from 127.0.0.1
-- Playing 'conf-onlyperson' (language 'en')
-- Executing AGI("SIP/69.94.226.174-0820b450", "agi://127.0.0.1:4577//call_log") in new stack
-- AGI Script agi://127.0.0.1:4577//call_log completed, returning 0
-- Executing Dial("SIP/69.94.226.174-0820b450", "sip/gs102|55|o") in new stack
Apr 20 11:32:15 NOTICE[25746]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/69.94.226.174-0820b450", "") in new stack
== Spawn extension (default, 8778880524, 3) exited non-zero on 'SIP/69.94.226.174-0820b450'
-- Executing DeadAGI("SIP/69.94.226.174-0820b450", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing AGI("SIP/69.94.226.174-08210990", "agi://127.0.0.1:4577//call_log") in new stack
-- AGI Script agi://127.0.0.1:4577//call_log completed, returning 0
-- Executing Dial("SIP/69.94.226.174-08210990", "sip/gs102|55|o") in new stack
Apr 20 11:32:20 NOTICE[25762]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/69.94.226.174-08210990", "") in new stack
== Spawn extension (default, 8778880524, 3) exited non-zero on 'SIP/69.94.226.174-08210990'
-- Executing DeadAGI("SIP/69.94.226.174-08210990", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing AGI("SIP/69.94.226.174-08215ed0", "agi://127.0.0.1:4577//call_log") in new stack
Apr 20 11:32:25 WARNING[25377]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x8202350', 9 retries!
-- AGI Script agi://127.0.0.1:4577//call_log completed, returning 0
-- Executing Dial("SIP/69.94.226.174-08215ed0", "sip/gs102|55|o") in new stack
Apr 20 11:32:25 NOTICE[25778]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/69.94.226.174-08215ed0", "") in new stack
== Spawn extension (default, 8778880524, 3) exited non-zero on 'SIP/69.94.226.174-08215ed0'
-- Executing DeadAGI("SIP/69.94.226.174-08215ed0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing AGI("SIP/69.94.226.174-0821b410", "agi://127.0.0.1:4577//call_log") in new stack
-- AGI Script agi://127.0.0.1:4577//call_log completed, returning 0
-- Executing Dial("SIP/69.94.226.174-0821b410", "sip/gs102|55|o") in new stack
Apr 20 11:32:30 NOTICE[25791]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/69.94.226.174-0821b410", "") in new stack
== Spawn extension (default, 8778880524, 3) exited non-zero on 'SIP/69.94.226.174-0821b410'
-- Executing DeadAGI("SIP/69.94.226.174-0821b410", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing AGI("SIP/69.94.226.174-08205f10", "agi://127.0.0.1:4577//call_log") in new stack
Apr 20 11:32:43 WARNING[25377]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x8202350', 9 retries!
-- AGI Script agi://127.0.0.1:4577//call_log completed, returning 0
-- Executing Dial("SIP/69.94.226.174-08205f10", "sip/gs102|55|o") in new stack
Apr 20 11:32:43 NOTICE[25832]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/69.94.226.174-08205f10", "") in new stack
== Spawn extension (default, 8778880524, 3) exited non-zero on 'SIP/69.94.226.174-08205f10'
-- Executing DeadAGI("SIP/69.94.226.174-08205f10", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing AGI("SIP/69.94.226.174-0820b450", "agi://127.0.0.1:4577//call_log") in new stack
-- AGI Script agi://127.0.0.1:4577//call_log completed, returning 0
-- Executing Dial("SIP/69.94.226.174-0820b450", "sip/gs102|55|o") in new stack
Apr 20 11:32:52 NOTICE[25861]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/69.94.226.174-0820b450", "") in new stack
== Spawn extension (default, 8778880524, 3) exited non-zero on 'SIP/69.94.226.174-0820b450'
-- Executing DeadAGI("SIP/69.94.226.174-0820b450", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0



Please advice...thanks