> ow are you getting SIP-related errors from Console/DSP? Posting a > console log would be most helpful, as many people on the mailing list > are not telepathic :-) > > -- > Kevin P. Fleming > Director of Software Technologies > Digium, Inc. - "The Genuine Asterisk Experience" (TM)
Kevin, below is the log your talking about. please note no configuration files were changed from 1.4 to 1.6, going back to 1.4 works again. Jerry ---------------------- Asterisk 1.6.0-beta9, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <[EMAIL PROTECTED]> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': == Found [0;37;40m[1;30;40m == [0;37;40mParsing '/etc/asterisk/extconfig.conf': [1;30;40m == [0;37;40mFound [0mConnected to Asterisk 1.6.0-beta9 currently running on ebox4300 (pid = 4877) ebox4300*CLI> Verbosity is at least 5 [Kebox4300*CLI> <--- SIP read from UDP://192.168.1.8:5060 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;rport From: "Jerry Geis 204" <sip:[EMAIL PROTECTED]>;tag=as7d1f7b71 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 Jul 2008 16:53:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 20475 20475 IN IP4 192.168.1.8 s=session c=IN IP4 192.168.1.8 t=0 0 m=audio 14322 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> �--- (14 headers 14 lines) --- � == Using SIP RTP CoS mark 5 � == Using SIP VRTP CoS mark 6 �Sending to 192.168.1.8 : 5060 (NAT) �Using INVITE request as basis request - [EMAIL PROTECTED] �No user '3175661677' in SIP users list �Found peer 'devcentos5x64_to_ebox4300' for '3175661677' from 192.168.1.8:5060 � <--- Reliably Transmitting (no NAT) to 192.168.1.8:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;received=192.168.1.8;rport=5060 From: "Jerry Geis 204" <sip:[EMAIL PROTECTED]>;tag=as7d1f7b71 To: <sip:[EMAIL PROTECTED]>;tag=as324df4b6 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0-beta9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0e961d2a" Content-Length: 0 <------------> �Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) � [Kebox4300*CLI> <--- SIP read from UDP://192.168.1.8:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK029ea409;rport From: "Jerry Geis 204" <sip:[EMAIL PROTECTED]>;tag=as7d1f7b71 To: <sip:[EMAIL PROTECTED]>;tag=as324df4b6 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> �--- (10 headers 0 lines) --- � <--- SIP read from UDP://192.168.1.8:5060 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6a460d62;rport From: "Jerry Geis 204" <sip:[EMAIL PROTECTED]>;tag=as7d1f7b71 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="devcentos5x64_to_ebox4300", realm="asterisk", algorithm=MD5, uri="sip:[EMAIL PROTECTED]", nonce="0e961d2a", response="1a8e257ae008af4156b1f65be8d4d267" Date: Mon, 21 Jul 2008 16:53:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 20475 20476 IN IP4 192.168.1.8 s=session c=IN IP4 192.168.1.8 t=0 0 m=audio 14322 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> �--- (15 headers 14 lines) --- �Sending to 192.168.1.8 : 5060 (NAT) �Using INVITE request as basis request - [EMAIL PROTECTED] �No user '3175661677' in SIP users list �Found peer 'devcentos5x64_to_ebox4300' for '3175661677' from 192.168.1.8:5060 �Found RTP audio format 0 �Found RTP audio format 8 �Found RTP audio format 3 �Found RTP audio format 101 �Peer audio RTP is at port 192.168.1.8:14322 �Found audio description format PCMU for ID 0 �Found audio description format PCMA for ID 8 �Found audio description format GSM for ID 3 �Found audio description format telephone-event for ID 101 �Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) �Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) �Peer audio RTP is at port 192.168.1.8:14322 �Looking for mediaport_audio_visual in smvoice-mediaport (domain 192.168.1.25) � <--- Reliably Transmitting (no NAT) to 192.168.1.8:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6a460d62;received=192.168.1.8;rport=5060 From: "Jerry Geis 204" <sip:[EMAIL PROTECTED]>;tag=as7d1f7b71 To: <sip:[EMAIL PROTECTED]>;tag=as324df4b6 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0-beta9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> �[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite: Call from 'devcentos5x64_to_ebox4300' to extension 'mediaport_audio_visual' rejected because extension not found. �Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) � <--- SIP read from UDP://192.168.1.8:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6a460d62;rport From: "Jerry Geis 204" <sip:[EMAIL PROTECTED]>;tag=as7d1f7b71 To: <sip:[EMAIL PROTECTED]>;tag=as324df4b6 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> �--- (10 headers 0 lines) --- � [Kebox4300*CLI> quit Executing last minute cleanups Asterisk cleanly ending (0). [0m _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
