[resolve] Call Failed : Reason Code 503 on my Ip Phone

lordaker · March 15, 2018, 4:26pm

I have this error message when I try to make an external call :

Executing [042680871@from-internal:1] Dial("PJSIP/102-00000004", "PJSIP/@belgium-voip-provider") in new stack
   -- Called PJSIP/@belgium-voip-provider
 == Everyone is busy/congested at this time (1:0/0/1)
   -- Auto fallthrough, channel 'PJSIP/102-00000004' status is 'CHANUNAVAIL'

This is my extensions.conf :

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=DAHDI/G2
TRUNKMSD=1

[from-internal]
exten => 101,1,NoOp(First Line)
same => n,Dial(PJSIP/${exten})
same => n,VoiceMail(101,u)
same => n, Hangup()

exten => 102,1,NoOp(First Line)
same => n,Dial(PJSIP/${exten})
same => n,VoiceMail(101,u)
same => n, Hangup()

exten => _XXXX,1,Dial(PJSIP/${exten}@belgium-voip-provider)
exten => _XXXXXXXXX,1,Dial(PJSIP/${exten}@belgium-voip-provider)
exten => _XXXXXXXXXX,1,Dial(PJSIP/${exten}@belgium-voip-provider)

jcolp · March 15, 2018, 4:27pm

It’s ${EXTEN} not ${exten}

lordaker · March 15, 2018, 4:33pm

Ok, I change the text in extensions.conf and reload the diaplan but nothing have changed.

jcolp · March 15, 2018, 4:34pm

Please provide the new console output. You will also need to provide the pjsip.conf configuration and the output of “pjsip set logger on” for a call attempt.

lordaker · March 15, 2018, 5:07pm

This is my output from the Asterisk CLI :

Connected to Asterisk 13.19.2 currently running on IPBX-Asterisk (pid = 24220)
  == Setting global variable 'SIPDOMAIN' to '192.168.40.55'
    -- Executing [042680871@from-internal:1] Dial("PJSIP/102-0000000f", "PJSIP/042680871@belgium-voip-provider") in new stack
    -- Called PJSIP/042680871@belgium-voip-provider
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/102-0000000f' status is 'CHANUNAVAIL'

This is my pjsip.conf

;--
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements start
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[general]
udpbindaddr = 0.0.0.0:5060

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements end
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
--;

;================================ TRANSPORT
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:5060
external_media_address=My_Public_address
external_signaling_address=My_Public_address
allow_reload=yes
local_net=192.168.40.0/24


;============================ EXTENSION 102
[102]
type = aor
max_contacts = 1

[102]
type = auth
username = 102
password = 102

[102]
type = endpoint
context = from-internal
disallow = all
allow = ulaw,alaw
mailboxes = 102@default
auth = 102
outbound_auth = 102
aors = 102


;============================ EXTENSION 101
[101]
type = aor
max_contacts = 1

[101]
type = auth
username = 101
password = 101

[101]
type = endpoint
context = from-internal
disallow = all
allow = ulaw,alaw
mailboxes = 101@default
auth = 101
outbound_auth = 101
aors = 101

Code of my provider SIP:

lordaker · March 15, 2018, 7:20pm

I am actually registered to service provider.

IPBX-Asterisk*CLI> pjsip show registrations

 <Registration/ServerURI..............................>  <Auth..........>  <Status.......>
==========================================================================================

 belgium-voip-provider/sip:voip.belgium-voip.com:5060    belgium-voip-provider-auth  Registered

Objects found: 1

jcolp · March 15, 2018, 7:22pm

Registration does not impact outgoing calls usually. It just tells the remote side where you can be reached. You’ll need to provide the output of “pjsip set logger on” with a call attempt.

lordaker · March 15, 2018, 7:24pm

I have nothing when I type the command pjsip set logger

IPBX-Asterisk*CLI> pjsip set logger
No such command 'pjsip set logger ' (type 'core show help pjsip set logger' for other possible commands)

jcolp · March 15, 2018, 7:27pm

It’s “pjsip set logger on” not “pjsip set logger”.

lordaker · March 15, 2018, 7:28pm

Yes is “On” now and I Receive SIP response.
The differents informations are correct.

jcolp · March 15, 2018, 8:09pm

Can you provide it so it can be examined?

lordaker · March 16, 2018, 9:33am

 Setting global variable 'SIPDOMAIN' to '192.168.40.55'
    -- Executing [042680871@from-internal:1] Dial("PJSIP/102-00000002", "PJSIP/042680871@belgium-voip-provider") in new stack
    -- Called PJSIP/042680871@belgium-voip-provider
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/102-00000002' status is 'CHANUNAVAIL'

I’m connected to my voip provider. When I go to the portal of Belgium VoIP, I’ve the status connect to this account. I think the probleme is my dialplan configuration.

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
IAXINFO=guest                                   ; IAXtel username/password
TRUNK=DAHDI/G2                                  ; Trunk interface
TRUNKMSD=1                                      ; MSD digits to strip (usually 1 or 0)

;===================== Internal Calls ========================
[from-internal]
exten => _10X,1,NoOp(Lancement de l'appel)
same => n,Dial(PJSIP/${EXTEN})
same => n,VoiceMail(101,u)
same => n,Hangup()
same => n,NoOp(Fin de l'appel)


exten => 103,1,Answer()
same => n,Wait(1)
same => n,Playback(room-service)
same => n,Hangup()
same => n,NoOp(Fin de l'appel)


exten => _XXXX,1,Dial(PJSIP/${EXTEN}@belgium-voip-provider)
exten => _XXXXXXXXX,1,Dial(PJSIP/${EXTEN}@belgium-voip-provider)
exten => _XXXXXXXXXX,1,Dial(PJSIP/${EXTEN}@belgium-voip-provider)


;====================== Outgoing Calls ==========================
[from-belgium-voip]
exten => 003242771412,1,NoOp(Lancement de l'appel externe)
same => n,Dial(PJSIP/${EXTEN},20)
same => n,Handup()
same => n,NoOp(Fin de l'appel externe)

david551 · March 16, 2018, 9:36am

There is no evidence of a dialplan problem and you haven’t provided the protocol logging requested.

lordaker · March 16, 2018, 9:55am

jcolp · March 16, 2018, 10:00am

Is that during a call attempt? It doesn’t appear so, as the console output doesn’t show a call.

lordaker · March 16, 2018, 10:05am

Yes, it is during a call attempt. The console output show this in the console CLI :

Setting global variable 'SIPDOMAIN' to '192.168.40.55'
    -- Executing [042680871@from-internal:1] Dial("PJSIP/102-00000002", "PJSIP/042680871@belgium-voip-provider") in new stack
    -- Called PJSIP/042680871@belgium-voip-provider
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/102-00000002' status is 'CHANUNAVAIL'

And on the screen of phone ip I have this :

Call Failed
Reason Code : 503

jcolp · March 16, 2018, 10:07am

The image of the console you posted did NOT include what you’ve provided above. There needs to be both to understand what is going on and narrow it down.

jcolp · March 16, 2018, 10:08am

The output of “pjsip show endpoint belgium-voip-provider” and “pjsip show contacts” from the console would also be good.

lordaker · March 16, 2018, 10:09am

I try make some call in belgium because my provider is in Belgium.

(+32) 04 XXX XX XX for the Fix phone
(+32) 04 XXX XXX XX for the mobile

lordaker · March 16, 2018, 10:19am

Fisrt Part

Second Part:

Topic		Replies	Views
How do I forward to my voip adaptor Asterisk Support	4	237	September 28, 2009
Forcing a 503 error Asterisk Support	4	1463	February 5, 2007
Incoming calls don't work, why? Asterisk Support	30	4802	April 16, 2018
Error 503 after one outgoing call Asterisk SIP	13	7608	February 4, 2017
Resolving issue Asterisk SIP	7	1432	October 31, 2020

[resolve] Call Failed : Reason Code 503 on my Ip Phone

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