Cannot make inbound call

[elsayed.mohamed]

Hello everyone,
I am new to this forum and to VICIdial as well. I installed a vicidial server in my home network. I got a lot working. I can dial extension-to-extension and I can dial outbound calls too. However, I am not able to make inbound calls. the famous "sip/2.0 401-unauthorized" error. I changed several parameters in sip.conf, extensions.conf, the web interface, and my sip trunking configuration but no luck. I really appreciate your help.

Here is some information about my system. I can post/email any information if needed. Thanks.

Asterisk 13.34.0-vici currently running
****************************************************************************************************************************************
[CloudVision]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
type=friend
username=xxxxxxxxxx
secret=yyyyyyyy
host=w.x.y.z
dtmfmode=rfc2833
context=trunkinbound
****************************************************************************************************************************************
exten => _XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXX,2,Dial(SIP/CloudVision/${EXTEN},,tTo)
exten => _XXXXXXXXXX,4,Hangup()
****************************************************************************************************************************************
[general]
context=trunkinbound ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;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
;pedantic=yes ; Enable checking of tags in headers,
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
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
;compactheaders = yes ; send compact sip headers.
videosupport=no ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes ; generate manager events when sip ua
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes ; Turn on SIP debugging by default, from
;recordhistory=yes ; Record SIP history by default
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes ; Default false
;register => 1234:password@mysipprovider.com
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
externip = 96.56.237.2 ; Address that we're going to put in outbound SIP
;externhost=test.test.com ; Alternatively you can specify a domain
;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)
canreinvite=no ; Asterisk by default tries to redirect the
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes ; Save systemname in realtime database at registration
;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=mydomain.tld,mydomain-incoming
;domain=1.2.3.4 ; Add IP address as local domain
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
;autodomain=yes ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000
****************************************************************************************************************************************

[elsayed.mohamed]

<--- Reliably Transmitting (NAT) to 192.92.8.30:5061 --->
[Feb 22 21:22:26] SIP/2.0 401 Unauthorized
[Feb 22 21:22:26] Via: SIP/2.0/UDP 192.92.8.30:5061;branch=z9hG4bK-ldiyjdve5a5ikhsy;received=192.92.8.30;rport=5061
[Feb 22 21:22:26] From: <sip:17322593236@192.92.8.30>;tag=2yxbv35sokrmkmmz.o
[Feb 22 21:22:26] To: <sip:16408887676@96.56.237.2>;tag=as1b5b43d7
[Feb 22 21:22:26] Call-ID: 222565344_112764632@67.231.9.166
[Feb 22 21:22:26] CSeq: 417 INVITE
[Feb 22 21:22:26] Server: Asterisk PBX 13.34.0-vici
[Feb 22 21:22:26] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Feb 22 21:22:26] Supported: replaces, timer
[Feb 22 21:22:26] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5ff0a208"
[Feb 22 21:22:26] Content-Length: 0

[carpenox]

do you have your carrier setup to take inbound calls?

[elsayed.mohamed]

[url][/url]Hi Carpenox,
Thank you for trying to help me out.

I resolved the 401 Unauthorized message issue. Now the call goes through but I get the message "the number you have dialed is not in service" I am sure I have the number from my provider and it is configured. Below is the SIP exchange between my mobile (17322593236) and my SIP number (16408887676) I know the issue is with my [trunkinbound] section dialplan configuration in extensions.conf file. However, I am not able to configure the dialplan correctly.

exten => _XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXX,2,Set(CALLERID(num)=16408887676)
exten => _XXXXXXXXXX,3,Dial(SIP/CloudVision/${EXTEN},,tTo)
exten => _XXXXXXXXXX,4,Hangup()


Below is the SIP exchange of the call from my mobile to 16408887676.



[Feb 25 18|51|54] == Using SIP RTP CoS mark 5
[Feb 25 18|51|54] > 0x7fb9440aff40 -- Strict RTP learning after remote address set to| x.y.w.z|39228
[Feb 25 18|51|54] -- Executing [0204254938@trunkinbound|1] AGI("SIP|x.y.w.z-00000054", "agi-DID_route.agi") in new stack
[Feb 25 18|51|54] -- Launched AGI Script |usr|share|asterisk|agi-bin|agi-DID_route.agi
[Feb 25 18|51|54] -- <SIP|x.y.w.z-00000054>AGI Script agi-DID_route.agi completed, returning 0
[Feb 25 18|51|54] -- Executing [9998811112@default|1] Wait("SIP|x.y.w.z-00000054", "2") in new stack
[Feb 25 18|51|56] -- Executing [9998811112@default|2] Answer("SIP|x.y.w.z-00000054", "") in new stack
[Feb 25 18|51|57] -- Executing [9998811112@default|3] Playback("SIP|x.y.w.z-00000054", "ss-noservice") in new stack
[Feb 25 18|51|57] -- <SIP|x.y.w.z-00000054> Playing 'ss-noservice.gsm' (language 'en')
[Feb 25 18|51|57] > 0x7fb9440aff40 -- Strict RTP switching to RTP target address x.y.w.z|39228 as source
[Feb 25 18|51|59] NOTICE[2578]| chan_sip.c|15842 sip_reregister| -- Re-registration for 0204254938@xyx.net

[carpenox]

Your DID is routed to EXTEN which you have set to the "no service" sound. Change the routing to however you want the call sent to:

[elsayed.mohamed]

It is already routed to extension 100. Am I missing something? I am not able to attach a screen show to show you.