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Hi, i have a little problem with GUI and Sip Trunk.<br><br>I have set up my asterisk server 1.6.0.6 with GUI 2.0<br><br>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.<br><br><-------------><br><--- SIP read from UDP://83.211.227.21:5060 ---><br>INVITE sip:s@150.217.13.132 SIP/2.0<br>Record-Route: <sip:83.211.227.21;ftag=2713899C-671;lr=on><br>Record-Route: <sip:83.211.227.13;ftag=2713899C-671;lr=on><br>Via: SIP/2.0/UDP 83.211.227.21;branch=0<br>Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK02bd.ae53af21.0<br>Via: SIP/2.0/UDP 83.211.2.218:5060;rport=56083;x-route-tag="tgrp:Slot6";branch=z9hG4bK3B6A910B<br>From: <sip:347*******@83.211.2.218>;tag=2713899C-671<br>To: <sip:0574******@voip.eutelia.it><br>Call-ID: 4036BFDE-40A11DE-BA12BD79-214756FC@83.211.2.218<br>CSeq: 102 INVITE<br>Max-Forwards: 8<br>Remote-Party-ID: <sip:347*******@83.211.2.218>;party=calling;screen=yes;privacy=off<br>Contact: <sip:347*******@83.211.2.218:5060><br>Expires: 180<br>Content-Type: application/sdp<br>Content-Length: 415<br>v=0<br>o=CiscoSystemsSIP-GW-UserAgent 9289 1505 IN IP4 83.211.2.218<br>s=SIP Call<br>c=IN IP4 62.94.199.36<br>t=0 0<br>m=audio 63772 RTP/AVP 18 8 0 4 3 125 101<br>c=IN IP4 62.94.199.36<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=yes<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:4 G723/8000<br>a=fmtp:4 bitrate=5.3;annexa=no<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:125 X-CCD/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br><-------------><br>--- (16 headers 17 lines) ---<br> == Using SIP RTP CoS mark 5<br>Sending to 83.211.227.21 : 5060 (no NAT)<br>Using INVITE request as basis request - 4036BFDE-40A11DE-BA12BD79-214756FC@83.211.2.218<br>No user '347*******' in SIP users list<br>Found peer 'trunk_1' for '347*******' from 83.211.227.21:5060<br><--- Reliably Transmitting (no NAT) to 83.211.227.21:5060 ---><br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 83.211.227.21;branch=0;received=83.211.227.21<br>Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK02bd.ae53af21.0<br>Via: SIP/2.0/UDP 83.211.2.218:5060;rport=56083;x-route-tag="tgrp:Slot6";branch=z9hG4bK3B6A910B<br>From: <sip:347*******@83.211.2.218>;tag=2713899C-671<br>To: <sip:0574******@voip.eutelia.it>;tag=as415e84f8<br>Call-ID: 4036BFDE-40A11DE-BA12BD79-214756FC@83.211.2.218<br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4df0ff3d"<br>Content-Length: 0<br><------------><br>Scheduling destruction of SIP dialog '4036BFDE-40A11DE-BA12BD79-214756FC@83.211.2.218' in 32000 ms (Method: INVITE)<br><--- SIP read from UDP://83.211.227.21:5060 ---><br>ACK sip:s@150.217.13.132 SIP/2.0<br>Max-Forwards: 15<br>Record-Route: <sip:83.211.227.21;ftag=2713899C-671;lr=on><br>Via: SIP/2.0/UDP 83.211.227.21;branch=0<br>Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK02bd.ae53af21.0<br>From: <sip:347*******@83.211.2.218>;tag=2713899C-671<br>Call-ID: 4036BFDE-40A11DE-BA12BD79-214756FC@83.211.2.218<br>To: <sip:0574******@voip.eutelia.it>;tag=as415e84f8<br>CSeq: 102 ACK<br>Content-Length: 0<br><-------------><br>--- (10 headers 0 lines) ---<br><--- SIP read from UDP://83.211.227.21:5060 ---><br>INVITE sip:s@150.217.13.132 SIP/2.0<br>Record-Route: <sip:83.211.227.21;ftag=23BDC0D0-26A9;lr=on><br>Record-Route: <sip:83.211.227.13;ftag=23BDC0D0-26A9;lr=on><br>Via: SIP/2.0/UDP 83.211.227.21;branch=0<br>Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bKe114.c9f5b0a6.0<br>Via: SIP/2.0/UDP 62.94.71.96:5060;rport=52353;x-route-tag="tgrp:Slot7";branch=z9hG4bK45FF31981<br>From: <sip:347*******@62.94.71.96>;tag=23BDC0D0-26A9<br>To: <sip:0574******@voip.eutelia.it><br>Call-ID: 4074D407-40A11DE-84A49CF3-13CC2153@62.94.71.96<br>CSeq: 102 INVITE<br>Max-Forwards: 8<br>Remote-Party-ID: <sip:347*******@62.94.71.96>;party=calling;screen=yes;privacy=off<br>Contact: <sip:347*******@62.94.71.96:5060><br>Expires: 180<br>Content-Type: application/sdp<br>Content-Length: 434<br>v=0<br>o=CiscoSystemsSIP-GW-UserAgent 6954 399 IN IP4 62.94.71.96<br>s=SIP Call<br>c=IN IP4 62.94.199.37<br>t=0 0<br>m=audio 62482 RTP/AVP 18 8 0 4 3 125 101<br>c=IN IP4 62.94.199.37<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=yes<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:4 G723/8000<br>a=fmtp:4 bitrate=5.3;annexa=no<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:125 X-CCD/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=direction:passive<br><-------------><br>--- (16 headers 18 lines) ---<br> == Using SIP RTP CoS mark 5<br>Sending to 83.211.227.21 : 5060 (no NAT)<br>Using INVITE request as basis request - 4074D407-40A11DE-84A49CF3-13CC2153@62.94.71.96<br>No user '347*******' in SIP users list<br>Found peer 'trunk_1' for '347*******' from 83.211.227.21:5060<br><--- Reliably Transmitting (no NAT) to 83.211.227.21:5060 ---><br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 83.211.227.21;branch=0;received=83.211.227.21<br>Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bKe114.c9f5b0a6.0<br>Via: SIP/2.0/UDP 62.94.71.96:5060;rport=52353;x-route-tag="tgrp:Slot7";branch=z9hG4bK45FF31981<br>From: <sip:347*******@62.94.71.96>;tag=23BDC0D0-26A9<br>To: <sip:0574******@voip.eutelia.it>;tag=as1d4d87a1<br>Call-ID: 4074D407-40A11DE-84A49CF3-13CC2153@62.94.71.96<br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4b39d7fa"<br>Content-Length: 0<br><------------><br>Scheduling destruction of SIP dialog '4074D407-40A11DE-84A49CF3-13CC2153@62.94.71.96' in 32000 ms (Method: INVITE)<br><--- SIP read from UDP://83.211.227.21:5060 ---><br>ACK sip:s@150.217.13.132 SIP/2.0<br>Max-Forwards: 15<br>Record-Route: <sip:83.211.227.21;ftag=23BDC0D0-26A9;lr=on><br>Via: SIP/2.0/UDP 83.211.227.21;branch=0<br>Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bKe114.c9f5b0a6.0<br>From: <sip:347*******@62.94.71.96>;tag=23BDC0D0-26A9<br>Call-ID: 4074D407-40A11DE-84A49CF3-13CC2153@62.94.71.96<br>To: <sip:0574******@voip.eutelia.it>;tag=as1d4d87a1<br>CSeq: 102 ACK<br>Content-Length: 0<br><-------------><br><br><br>This is a part of my configuration...<br><br>users.conf<br><br>[trunk_1]<br>context=DID_trunk_1<br>host=voip.eutelia.it<br>username=0574******<br>insecure=no<br>secret=********<br>trunkname=eutelia<br>hasiax=no<br>registeriax=no<br>hassip=yes<br>registersip=yes<br>trunkstyle=voip<br>hasexten=no<br>disallow=all<br>allow=all<br><br>extensions.conf<br><br>[default]<br><br>[DLPN_DialPlan1]<br>include = default<br>include = ringgroups<br><br>[DID_trunk_1]<br>include = DID_trunk_1_timeinterval_all,${timeinterval_all}<br>include = DID_trunk_1_default<br><br>[DID_trunk_1_default]<br><br>[DID_trunk_1_timeinterval_all]<br>exten = _X.,1,Goto(default,6000,1)<br><br><br>Giovanni Giusti.<br><br /><hr />Scoprilo insieme ai nuovi servizi Windows Live! <a href='http://download.live.com/messenger/?mkt=it-it' target='_new'>Messenger 9: oltre le parole.</a></body>
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