[asterisk-gui] Incoming Calling Rules - Trunk Sip
Giovanni Giusti
d2_won at hotmail.it
Fri Feb 27 09:16:52 CST 2009
Hi, i have a little problem with GUI and Sip Trunk.
I have set up my asterisk server 1.6.0.6 with GUI 2.0
When i call outbound OK, but when i call with my cellphone (347*******) the number of sip trunk(0574******), i recevie e fast hangup and in Debug console i see this log file.
<------------->
<--- SIP read from UDP://83.211.227.21:5060 --->
INVITE sip:s at 150.217.13.132 SIP/2.0
Record-Route: <sip:83.211.227.21;ftag=2713899C-671;lr=on>
Record-Route: <sip:83.211.227.13;ftag=2713899C-671;lr=on>
Via: SIP/2.0/UDP 83.211.227.21;branch=0
Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK02bd.ae53af21.0
Via: SIP/2.0/UDP 83.211.2.218:5060;rport=56083;x-route-tag="tgrp:Slot6";branch=z9hG4bK3B6A910B
From: <sip:347*******@83.211.2.218>;tag=2713899C-671
To: <sip:0574******@voip.eutelia.it>
Call-ID: 4036BFDE-40A11DE-BA12BD79-214756FC at 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: 415
v=0
o=CiscoSystemsSIP-GW-UserAgent 9289 1505 IN IP4 83.211.2.218
s=SIP Call
c=IN IP4 62.94.199.36
t=0 0
m=audio 63772 RTP/AVP 18 8 0 4 3 125 101
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
<------------->
--- (16 headers 17 lines) ---
== Using SIP RTP CoS mark 5
Sending to 83.211.227.21 : 5060 (no NAT)
Using INVITE request as basis request - 4036BFDE-40A11DE-BA12BD79-214756FC at 83.211.2.218
No user '347*******' in SIP users list
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=z9hG4bK02bd.ae53af21.0
Via: SIP/2.0/UDP 83.211.2.218:5060;rport=56083;x-route-tag="tgrp:Slot6";branch=z9hG4bK3B6A910B
From: <sip:347*******@83.211.2.218>;tag=2713899C-671
To: <sip:0574******@voip.eutelia.it>;tag=as415e84f8
Call-ID: 4036BFDE-40A11DE-BA12BD79-214756FC at 83.211.2.218
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4df0ff3d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '4036BFDE-40A11DE-BA12BD79-214756FC at 83.211.2.218' in 32000 ms (Method: INVITE)
<--- SIP read from UDP://83.211.227.21:5060 --->
ACK sip:s at 150.217.13.132 SIP/2.0
Max-Forwards: 15
Record-Route: <sip:83.211.227.21;ftag=2713899C-671;lr=on>
Via: SIP/2.0/UDP 83.211.227.21;branch=0
Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK02bd.ae53af21.0
From: <sip:347*******@83.211.2.218>;tag=2713899C-671
Call-ID: 4036BFDE-40A11DE-BA12BD79-214756FC at 83.211.2.218
To: <sip:0574******@voip.eutelia.it>;tag=as415e84f8
CSeq: 102 ACK
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP://83.211.227.21:5060 --->
INVITE sip:s at 150.217.13.132 SIP/2.0
Record-Route: <sip:83.211.227.21;ftag=23BDC0D0-26A9;lr=on>
Record-Route: <sip:83.211.227.13;ftag=23BDC0D0-26A9;lr=on>
Via: SIP/2.0/UDP 83.211.227.21;branch=0
Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bKe114.c9f5b0a6.0
Via: SIP/2.0/UDP 62.94.71.96:5060;rport=52353;x-route-tag="tgrp:Slot7";branch=z9hG4bK45FF31981
From: <sip:347*******@62.94.71.96>;tag=23BDC0D0-26A9
To: <sip:0574******@voip.eutelia.it>
Call-ID: 4074D407-40A11DE-84A49CF3-13CC2153 at 62.94.71.96
CSeq: 102 INVITE
Max-Forwards: 8
Remote-Party-ID: <sip:347*******@62.94.71.96>;party=calling;screen=yes;privacy=off
Contact: <sip:347*******@62.94.71.96:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 434
v=0
o=CiscoSystemsSIP-GW-UserAgent 6954 399 IN IP4 62.94.71.96
s=SIP Call
c=IN IP4 62.94.199.37
t=0 0
m=audio 62482 RTP/AVP 18 8 0 4 3 125 101
c=IN IP4 62.94.199.37
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 CoS mark 5
Sending to 83.211.227.21 : 5060 (no NAT)
Using INVITE request as basis request - 4074D407-40A11DE-84A49CF3-13CC2153 at 62.94.71.96
No user '347*******' in SIP users list
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=z9hG4bKe114.c9f5b0a6.0
Via: SIP/2.0/UDP 62.94.71.96:5060;rport=52353;x-route-tag="tgrp:Slot7";branch=z9hG4bK45FF31981
From: <sip:347*******@62.94.71.96>;tag=23BDC0D0-26A9
To: <sip:0574******@voip.eutelia.it>;tag=as1d4d87a1
Call-ID: 4074D407-40A11DE-84A49CF3-13CC2153 at 62.94.71.96
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4b39d7fa"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '4074D407-40A11DE-84A49CF3-13CC2153 at 62.94.71.96' in 32000 ms (Method: INVITE)
<--- SIP read from UDP://83.211.227.21:5060 --->
ACK sip:s at 150.217.13.132 SIP/2.0
Max-Forwards: 15
Record-Route: <sip:83.211.227.21;ftag=23BDC0D0-26A9;lr=on>
Via: SIP/2.0/UDP 83.211.227.21;branch=0
Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bKe114.c9f5b0a6.0
From: <sip:347*******@62.94.71.96>;tag=23BDC0D0-26A9
Call-ID: 4074D407-40A11DE-84A49CF3-13CC2153 at 62.94.71.96
To: <sip:0574******@voip.eutelia.it>;tag=as1d4d87a1
CSeq: 102 ACK
Content-Length: 0
<------------->
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)
Giovanni Giusti.
_________________________________________________________________
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http://livesearch.it.msn.com/
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