vicidial inbound rule

[ehermouet]

hi all

i can make outbound rule but impossible to make inbound

[Apr 17 09:20:18] VERBOSE[2698] logger.c: [Apr 17 09:20:18] --- (14 headers 17 lines) ---
[Apr 17 09:20:18] VERBOSE[2698] logger.c: [Apr 17 09:20:18] Sending to 192.168.1.42 : 5060 (NAT)
[Apr 17 09:20:18] VERBOSE[2698] logger.c: [Apr 17 09:20:18] Using INVITE request as basis request - YjYwNGE4NWQ3NzhmZWNhMTc4ZTA4NTU2MDZhNDE3ZDY.
[Apr 17 09:20:18] VERBOSE[2698] logger.c: [Apr 17 09:20:18] Found peer 'gs102'
[Apr 17 09:20:18] WARNING[2698] chan_sip.c: username mismatch, have <gs102>, digest has <>
[Apr 17 09:20:18] NOTICE[2698] chan_sip.c: Failed to authenticate user "Informatique"<sip:131@192.168.1.42:5060>;tag=786a8433
[Apr 17 09:20:18] VERBOSE[2698] logger.c: [Apr 17 09:20:18]
<--- Reliably Transmitting (NAT) to 192.168.1.42:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-d8754z-2119345f2705cc21-1---d8754z-;received=192.168.1.42;rport=5060
From: "Informatique"<sip:131@192.168.1.42:5060>;tag=786a8433
To: <sip:500@192.168.1.42>;tag=as31cf0697
Call-ID: YjYwNGE4NWQ3NzhmZWNhMTc4ZTA4NTU2MDZhNDE3ZDY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0



my carrer config is

[3cx500]
disallow=all
allow=ulaw
type=friend
context=trunkinbound
username=500
secret=d2b93r9
host=192.168.1.42
dtmfmode=rfc2833
fromdomain=192.168.1.42
fromuser=500
insecure = invite

asterisk 1.4.44 vicidial 4.0.3

tks advance

[williamconley]

apparently your server at 1.42 is using "131" instead of "500" and likely a different password to try to make the inbound call.

[ehermouet]

Hi William

131 is my other extension on other IPBX server.
exactly how it works

IPBX 1
extension 100 to 500
500 is carrier on vicidial

Vicidial server
500 is carrier connected to IPBX 1
outbound call ok
not inbound from IPBX 1

i want to remove auth on icomming call
tks

[ehermouet]

I have add auth=no

but now i have


[Apr 18 05:54:26] VERBOSE[2698] logger.c: [Apr 18 05:54:26] Looking for s in trunkinbound (domain 192.168.1.41)
[Apr 18 05:54:26] VERBOSE[2698] logger.c: [Apr 18 05:54:26]
<--- Reliably Transmitting (NAT) to 192.168.1.42:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.42:5060;branch=z9hG4bK-d8754z-fa1c68693740be1b-1---d8754z-;received=192.168.1.42;rport=5060
From: "Informatique"<sip:131@192.168.1.42:5060>;tag=4065b90b
To: <sip:500@192.168.1.42>;tag=as7728532b
Call-ID: NWJkZGRkMzhmODExOTM4ZGNjMjIzNWYxMWJkNzViZjM.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Apr 18 05:54:26] NOTICE[2698] chan_sip.c: Call from '500' to extension 's' rejected because extension not found.



[trunkinbound]
; DID call routing process
; exten => _XXXXXXXXXX,1,AGI(agi-DID_route.agi) ; use this one instead of the one below if you are having delay issues, and match to number of received digits
exten => _X.,1,AGI(agi-DID_route.agi)

; If you have DIDs that arrive with a plus sign at the beginning then uncomment
;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})

[williamconley]

where did you add auth=no?

what number is being dialed? the system thinks "s" is being dialed, which could be after an earlier failure where the real number was dialed but you did not include that part ...

we've also experienced a few freeswitch and other servers that used non-standard sip methods to dial numbers, so we had to alter the incoming call to get the number being dialed.

for now, if you modify "[trunkinbound]" in "extensions.conf" and add:

Code: Select all

exten => s,1,NoOp(${EXTEN}|${CALLERID(NUM)})
exten => s,n,AGI(agi-DID_route.agi)

the "default" DID in Inbound->DIDs will be used.