Asterisk-Gui incoming calling rules not work

I have a little problem with incoming calling rules this is my configuration.

[code]Command>core show version

Asterisk SVN-branch-1.6.1-r176023 built by root @ giovanni-laptop on a i686 running Linux on 2009-02-16 10:26:19 UTC

GUI-version : SVN-branch-2.0-r4512
[/code]
The outbound call with trunk work,

[code]Command> sip show registry

Host dnsmgr Username Refresh State Reg.Time
voip.eutelia.it:5060 N 0574****** 105 Registered Tue, 17 Feb 2009 12:53:29
1 SIP registrations.[/code]

When i call the trunk number “0574******” with my mobilephone in Asterisk console i receive this sip debug

[code]<— SIP read from UDP://83.211.227.21:5060 —>
INVITE sip:s@150.217.13.132 SIP/2.0
Record-Route: sip:83.211.227.21;ftag=152D3C0-D9D;lr=on
Record-Route: sip:83.211.227.13;ftag=152D3C0-D9D;lr=on
Via: SIP/2.0/UDP 83.211.227.21;branch=0
Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK5caf.7b94e614.0
Via: SIP/2.0/UDP 83.211.2.218:5060;rport=58711;branch=z9hG4bK14E506624C
From: sip:347*******@83.211.2.218;tag=152D3C0-D9D
To: sip:0574******@voip.eutelia.it
Call-ID: FE3EA820-FC2011DD-9118C043-51088926@83.211.2.218
CSeq: 102 INVITE
Max-Forwards: 8
Remote-Party-ID: sip:347*******@83.211.2.218;party=calling;screen=yes;privacy=off
Contact: sip:347*******@83.211.2.218:5060
Expires: 180
Content-Type: application/sdp
Content-Length: 435

v=0
o=CiscoSystemsSIP-GW-UserAgent 4514 271 IN IP4 83.211.2.218
s=SIP Call
c=IN IP4 62.94.199.38
t=0 0
m=audio 63746 RTP/AVP 18 8 0 4 3 125 101
c=IN IP4 62.94.199.38
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=5.3;annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:125 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive

<------------->
— (16 headers 18 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 83.211.227.21 : 5060 (no NAT)
Using INVITE request as basis request - FE3EA820-FC2011DD-9118C043-51088926@83.211.2.218
Found peer ‘trunk_1’ for ‘347*******’ from 83.211.227.21:5060

<— Reliably Transmitting (no NAT) to 83.211.227.21:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 83.211.227.21;branch=0;received=83.211.227.21
Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK5caf.7b94e614.0
Via: SIP/2.0/UDP 83.211.2.218:5060;rport=58711;branch=z9hG4bK14E506624C
From: sip:347*******@83.211.2.218;tag=152D3C0-D9D
To: sip:0574******@voip.eutelia.it;tag=as684b737e
Call-ID: FE3EA820-FC2011DD-9118C043-51088926@83.211.2.218
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6ac9ddd1"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘FE3EA820-FC2011DD-9118C043-51088926@83.211.2.218’ in 32000 ms (Method: INVITE)

<— SIP read from UDP://83.211.227.21:5060 —>
ACK sip:s@150.217.13.132 SIP/2.0
Max-Forwards: 15
Record-Route: sip:83.211.227.21;ftag=152D3C0-D9D;lr=on
Via: SIP/2.0/UDP 83.211.227.21;branch=0
Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK5caf.7b94e614.0
From: sip:347*******@83.211.2.218;tag=152D3C0-D9D
Call-ID: FE3EA820-FC2011DD-9118C043-51088926@83.211.2.218
To: sip:0574******@voip.eutelia.it;tag=as684b737e
CSeq: 102 ACK
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP://83.211.227.21:5060 —>
INVITE sip:s@150.217.13.132 SIP/2.0
Record-Route: sip:83.211.227.21;ftag=19C88924-D14;lr=on
Record-Route: sip:83.211.227.13;ftag=19C88924-D14;lr=on
Via: SIP/2.0/UDP 83.211.227.21;branch=0
Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK00cc.291df4a4.0
Via: SIP/2.0/UDP 83.211.72.90:5060;rport=62778;x-route-tag=“tgrp:Slot6”;branch=z9hG4bK244A0A35
From: sip:347*******@83.211.72.90;tag=19C88924-D14
To: sip:0574******@voip.eutelia.it
Call-ID: FE76874F-FC2011DD-BE1FF0E7-D80EAD1E@83.211.72.90
CSeq: 102 INVITE
Max-Forwards: 8
Remote-Party-ID: sip:347*******@83.211.72.90;party=calling;screen=yes;privacy=off
Contact: sip:347*******@83.211.72.90:5060
Expires: 180
Content-Type: application/sdp
Content-Length: 460

v=0
o=CiscoSystemsSIP-GW-UserAgent 2977 1164 IN IP4 83.211.72.90
s=SIP Call
c=IN IP4 62.94.199.36
t=0 0
m=audio 64742 RTP/AVP 18 8 0 4 3 125 101 19
c=IN IP4 62.94.199.36
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=5.3;annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:125 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=direction:passive

<------------->
— (16 headers 19 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 83.211.227.21 : 5060 (no NAT)
Using INVITE request as basis request - FE76874F-FC2011DD-BE1FF0E7-D80EAD1E@83.211.72.90
Found peer ‘trunk_1’ for ‘347*******’ from 83.211.227.21:5060

<— Reliably Transmitting (no NAT) to 83.211.227.21:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 83.211.227.21;branch=0;received=83.211.227.21
Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK00cc.291df4a4.0
Via: SIP/2.0/UDP 83.211.72.90:5060;rport=62778;x-route-tag=“tgrp:Slot6”;branch=z9hG4bK244A0A35
From: sip:347*******@83.211.72.90;tag=19C88924-D14
To: sip:0574******@voip.eutelia.it;tag=as33ce11cb
Call-ID: FE76874F-FC2011DD-BE1FF0E7-D80EAD1E@83.211.72.90
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="19ef3f99"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘FE76874F-FC2011DD-BE1FF0E7-D80EAD1E@83.211.72.90’ in 32000 ms (Method: INVITE)

<— SIP read from UDP://83.211.227.21:5060 —>
ACK sip:s@150.217.13.132 SIP/2.0
Max-Forwards: 15
Record-Route: sip:83.211.227.21;ftag=19C88924-D14;lr=on
Via: SIP/2.0/UDP 83.211.227.21;branch=0
Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK00cc.291df4a4.0
From: sip:347*******@83.211.72.90;tag=19C88924-D14
Call-ID: FE76874F-FC2011DD-BE1FF0E7-D80EAD1E@83.211.72.90
To: sip:0574******@voip.eutelia.it;tag=as33ce11cb
CSeq: 102 ACK
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘6082d7c40cbf6856691b2d3e008e1d21@127.0.1.1’ Method: REGISTER

[/code]

Any solutions?

ggiusti.

Configure the right user ID and password in the phone, or in Asterisk, or disable authentication in Asterisk.

How to disable authentication in Asterisk? change insecure=no to isecure=very in Trunk Setup???

This is a part of my configuration…

users.conf

[trunk_1]
context=DID_trunk_1
host=voip.eutelia.it
username=0574******
insecure=no
secret=********
trunkname=eutelia
hasiax=no
registeriax=no
hassip=yes
registersip=yes
trunkstyle=voip
hasexten=no
disallow=all
allow=all

extensions.conf

[default]

[DLPN_DialPlan1]
include = default
include = ringgroups

[DID_trunk_1]
include = DID_trunk_1_timeinterval_all,${timeinterval_all}
include = DID_trunk_1_default

[DID_trunk_1_default]

[DID_trunk_1_timeinterval_all]
exten = _X.,1,Goto(default,6000,1)

I believe your problem is with the configuration, or rather lack of it, for 347*******.

I suspect the parameter you are thinking of is allowguest, in sip.conf.

Try

insecure=port,invite

Cheers.

Marco Bruni
www.marcobruni.net

i have found a partial solution…

if i change

[DID_trunk_1_timeinterval_all]
exten = _X.,1,Goto(ringroups-custom-1,s,1)

to

[DID_trunk_1_timeinterval_all]
include = ringroups-custom-1

the incoming call work, but i have lost the filter ‘_X.’